localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249918279782@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298763-1168941506@172.16.100.100 From: ;tag=8662 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709 Contact: sip:2356221453@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 9879 9879 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 46948 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249918279782@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298763-1168941506@172.16.100.100 (54) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=8662 (62) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3 520709 (85) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356221453@172.16.100.100:5060;user=phone (54) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9879 9879 IN IP4 10.57.2.121 (41) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 46948 RTP/AVP 18 101 13 (31) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298763-1168941506@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298763-1168941506@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:46948 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.121:46948 [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249918279782 in default (domain 10.100.20.11) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356221453@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09ec46e0: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:26] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09ec46e0 [Jan 16 10:58:26] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:26] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:26] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:26] DEBUG[18357]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0264249918279782@default:1] Dial("SIP/5060-09ec46e0", "SIP/249918279782@196.29.163.6||t") in new stack [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:26] DEBUG[18357]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0264249918279782-1. [Jan 16 10:58:26] DEBUG[18357]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:26] DEBUG[18357]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:26] DEBUG[18357]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:2804 sip_call: Outgoing Call for 249918279782 [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 16840 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249918279782@196.29.163.6 SIP/2.0 (44) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport (64) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 2: From: "2356221453" ;tag=as7b0b0b1b (64) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 (55) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:26 GMT (35) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: m=audio 16840 RTP/AVP 18 (24) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249918279782@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport From: "2356221453" ;tag=as7b0b0b1b To: Contact: Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16840 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:26] DEBUG[18357]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #159 -- Called 249918279782@196.29.163.6 [Jan 16 10:58:26] DEBUG[18353]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:26] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356221453" ;tag=as7b0b0b1b To: CSeq: 102 INVITE Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356221453" ;tag=as7b0b0b1b (64) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 (55) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport=5060 (69) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #159 - INVITE (got response) [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '45e6c4827c0d0bc80a9897f c074585cd@217.113.64.11' Request 102: Found [Jan 16 10:58:26] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:27] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK384ee09e To: ;tag=t1168941502-co1347 From: "31614387178" ;tag=as6540313a Call-ID: 2f98b16154c5333a77ab617248623177@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1347-CPO02626 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1459079989 1459079989 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43992 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK384ee09e (64) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941502-co1347 (58) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "31614387178" ;tag=as6540313a (66) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 2f98b16154c5333a77ab617248623177@217.113.64.11 (55) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1347-CPO02626 (61) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1459079989 1459079989 IN IP4 216.226.69.244 (47) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43992 RTP/AVP 18 101 (28) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2f98b16154c5333a77ab617 248623177@217.113.64.11' Request 102: Found [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43992 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0a0596d8 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43992 [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:27] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-0a0596d8 is making progress passing it to SIP/5060-0a047990 [Jan 16 10:58:27] DEBUG[18354]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:27] DEBUG[18354]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:27] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:27] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:27] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:27] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:27] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:27] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:28] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:28] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK384ee09e To: ;tag=t1168941502-co1347 From: "31614387178" ;tag=as6540313a Call-ID: 2f98b16154c5333a77ab617248623177@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1347-CPO02626 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1459079989 1459079989 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43992 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK384ee09e (64) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941502-co1347 (58) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "31614387178" ;tag=as6540313a (66) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 2f98b16154c5333a77ab617248623177@217.113.64.11 (55) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1347-CPO02626 (61) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1459079989 1459079989 IN IP4 216.226.69.244 (47) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43992 RTP/AVP 18 101 (28) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '2f98b16154c5333a77ab617248623177@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43992 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0a0596d8 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43992 [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 2f98b16154c5333a77ab617248623177@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2207730302@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK68673886;rport From: "31614387178" ;tag=as6540313a To: ;tag=t1168941502-co1347 Contact: Call-ID: 2f98b16154c5333a77ab617248623177@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:28] DEBUG[18354]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0a0596d8 -- SIP/216.226.69.244-0a0596d8 answered SIP/5060-0a047990 [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:28] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:28] DEBUG[18354]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a047990 [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-0a047990 [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:28] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19440 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cbf-ac106464-3520700;received=172.16.100.100 From: ;tag=22774 To: ;tag=as03786025 Call-ID: 6994090221311298751-1168941501@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19440 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:28] DEBUG[18354]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #162 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432207730302@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298751-1168941501@172.16.100.100 From: ;tag=22774 To: ;tag=as03786025 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cbf-ac106464-3520700 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432207730302@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298751-1168941501@172.16.100.100 (54) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=22774 (64) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as03786025 (67) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cbf-ac106464-3 520700 (85) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #162 [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298751-1168941501@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:28] DEBUG[18354]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:28] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:28] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 404 Not Found From: "2356756282" ;tag=as588385c0 To: ;tag=e5c9caf1 CSeq: 102 INVITE Call-ID: 57bc8642285e30b93f9c29116f34da72@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK080dea51;rport=5060 Reason: Q.850;cause=1;text="Unallocated number" Content-Length: 0 <-------------> [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 404 Not Found (21) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356756282" ;tag=as588385c0 (64) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=e5c9caf1 (48) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 57bc8642285e30b93f9c29116f34da72@217.113.64.11 (55) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK080dea51;rport=5060 (69) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850;cause=1;text="Unallocated number" (47) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '57bc8642285e30b93f9c29116f34da72@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 404 to standard invite Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK080dea51;rport From: "2356756282" ;tag=as588385c0 To: ;tag=e5c9caf1 Contact: Call-ID: 57bc8642285e30b93f9c29116f34da72@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 57bc8642285e30b93f9c29116f34da72@217.113.64.11 -- SIP/196.29.163.6-09f02cd8 is circuit-busy [Jan 16 10:58:28] DEBUG[18356]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-09f02cd8' [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-09f02cd8, SIP callid 57bc8642285e30b93f9c29116f34da72@217.113.6 4.11) [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:3278 sip_hangup: update_call_counter(249734822305) - decrement call limit counter on hangup [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:28] DEBUG[18356]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-09f02cd8 == Everyone is busy/congested at this time (1:0/1/0) [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 16 10:58:28] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) [Jan 16 10:58:28] DEBUG[18356]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:28] DEBUG[18356]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. [Jan 16 10:58:28] DEBUG[18356]: pbx.c:1768 pbx_extension_helper: Launching 'Goto' -- Executing [0264249734822305@default:2] Goto("SIP/5060-09f01758", "s-CONGESTION|1") in new stack -- Goto (default,s-CONGESTION,1) [Jan 16 10:58:28] DEBUG[18356]: pbx.c:1768 pbx_extension_helper: Launching 'Hangup' -- Executing [s-CONGESTION@default:1] Hangup("SIP/5060-09f01758", "") in new stack [Jan 16 10:58:28] DEBUG[18356]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,s-CONGESTION,1) exited non-zero on 'SIP/5060-09f01758' == Spawn extension (default, s-CONGESTION, 1) exited non-zero on 'SIP/5060-09f01758' [Jan 16 10:58:28] DEBUG[18356]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:28] DEBUG[18356]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09f01758' [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09f01758, SIP callid 6994090221311298757-1168941504@172.16.100.100) [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311298757-1168941504@172.16.100.100' in 32000 ms (Method: INVITE) localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc5-ac106464-3520707;received=172.16.100.100 From: ;tag=7753 To: ;tag=as32cbfc04 Call-ID: 6994090221311298757-1168941504@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:58:28] DEBUG[18356]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #164 [Jan 16 10:58:28] DEBUG[18356]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09f01758 [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:28] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:28] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298757-1168941504@172.16.100.100 From: ;tag=7753 To: ;tag=as32cbfc04 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc5-ac106464-3520707 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298757-1168941504@172.16.100.100 (54) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=7753 (62) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as32cbfc04 (70) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc5-ac106464-3 520707 (85) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #164 [Jan 16 10:58:28] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298757-1168941504@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '57bc8642285e30b93f9c29116f34da72@217.113.64.11' Method: INVITE [Jan 16 10:58:28] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:28] DEBUG[18352]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:28] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:29] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:29] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:29] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:29] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:29] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:29] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:29] DEBUG[18353]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432209980909@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298745-1168941499@172.16.100.100 From: ;tag=21866 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb9-ac106464-3520698 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432209980909@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298745-1168941499@172.16.100.100 (54) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=21866 (65) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb9-ac106464-3 520698 (85) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298745-1168941499@172.16.100.100 localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb9-ac106464-3520698;received=172.16.100.100 From: ;tag=21866 To: ;tag=as3778f24a Call-ID: 6994090221311298745-1168941499@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Content-Length: 0 <------------> [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #165 localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb9-ac106464-3520698;received=172.16.100.100 From: ;tag=21866 To: ;tag=as3778f24a Call-ID: 6994090221311298745-1168941499@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:29] DEBUG[18353]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:29] DEBUG[18353]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-0a02e5c0' [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-0a02e5c0, SIP callid 70bc55591d2ff40e0aaf62d77c34e44e@217.113 .64.11) [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:3278 sip_hangup: update_call_counter(2209980909) - decrement call limit counter on hangup [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2209980909@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK509d88e2;rport From: "243851213877" ;tag=as3452116c To: Call-ID: 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #167 Scheduling destruction of SIP dialog '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:29] DEBUG[18353]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0a02e5c0 [Jan 16 10:58:29] DEBUG[18353]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:58:29] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:29] DEBUG[18353]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432209980909,1) exited non-zero on 'SIP/5060-09d967f8' == Spawn extension (default, 0432209980909, 1) exited non-zero on 'SIP/5060-09d967f8' [Jan 16 10:58:29] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:29] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:29] DEBUG[18353]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:29] DEBUG[18353]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09d967f8' [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09d967f8, SIP callid 6994090221311298745-1168941499@172.16.100.100) [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:29] DEBUG[18353]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:58:29] DEBUG[18353]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d967f8 [Jan 16 10:58:29] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:29] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:29] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432209980909@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298745-1168941499@172.16.100.100 From: ;tag=21866 To: ;tag=as3778f24a CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb9-ac106464-3520698 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432209980909@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298745-1168941499@172.16.100.100 (54) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=21866 (65) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as3778f24a (67) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb9-ac106464-3 520698 (85) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #165 [Jan 16 10:58:29] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298745-1168941499@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311298745-1168941499@172.16.100.100' Method: ACK [Jan 16 10:58:29] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK509d88e2 To: ;tag=t1168941500-co1338 From: "243851213877" ;tag=as3452116c Call-ID: 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1338-CPO02596 Content-Length: 0 <-------------> [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK509d88e2 (64) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941500-co1338 (58) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "243851213877" ;tag=as3452116c (68) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 (55) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1338-CPO02596 (61) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #167 [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' of Request 102: M atch Not Found localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK509d88e2 To: ;tag=t1168941500-co1338 From: "243851213877" ;tag=as3452116c Call-ID: 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1338-CPO02596 Content-Length: 0 <-------------> [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK509d88e2 (64) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941500-co1338 (58) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "243851213877" ;tag=as3452116c (68) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 (55) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1338-CPO02596 (61) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' of Request 102: M atch Found [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2209980909@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK509d88e2;rport From: "243851213877" ;tag=as3452116c To: ;tag=t1168941500-co1338 Contact: Call-ID: 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:30] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:30] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:30] DEBUG[18352]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:31] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:31] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:31] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205667787@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298767-1168941511@172.16.100.100 From: ;tag=27317 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3520714 Contact: sip:937654922@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3296 3296 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 30844 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205667787@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298767-1168941511@172.16.100.100 (54) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=27317 (62) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3 520714 (85) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:937654922@172.16.100.100:5060;user=phone (53) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3296 3296 IN IP4 10.57.3.22 (40) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 30844 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298767-1168941511@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298767-1168941511@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:30844 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.22:30844 [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205667787 in default (domain 10.100.20.11) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:937654922@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:31] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09d967f8: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3520714;received=172.16.100.100 From: ;tag=27317 To: Call-ID: 6994090221311298767-1168941511@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:31] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d967f8 [Jan 16 10:58:31] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:31] DEBUG[18358]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432205667787@default:1] Dial("SIP/5060-09d967f8", "SIP/2205667787@216.226.69.244||t") in new stack [Jan 16 10:58:31] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:31] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:31] DEBUG[18358]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432205667787-1. [Jan 16 10:58:31] DEBUG[18358]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:31] DEBUG[18358]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:31] DEBUG[18358]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:2804 sip_call: Outgoing Call for 2205667787 [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 10424 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205667787@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5d83a3d4;rport (64) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 2: From: "937654922" ;tag=as2aed489a (62) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 (55) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:31 GMT (35) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: m=audio 10424 RTP/AVP 18 (24) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205667787@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5d83a3d4;rport From: "937654922" ;tag=as2aed489a To: Contact: Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 10424 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:31] DEBUG[18358]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #169 -- Called 2205667787@216.226.69.244 [Jan 16 10:58:31] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:32] DEBUG[18352]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:32] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:32] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5d83a3d4 To: From: "937654922" ;tag=as2aed489a Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1395-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5d83a3d4 (64) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "937654922" ;tag=as2aed489a (62) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 (55) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1395-CPO00000 (58) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #169 - INVITE (got response) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0d006a9841c11388370adc3 b5e2c408b@217.113.64.11' Request 102: Found [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:32] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:32] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:32] DEBUG[18358]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:32] DEBUG[18358]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:32] DEBUG[18358]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14350 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:32] DEBUG[18358]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:32] DEBUG[18358]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3520714;received=172.16.100.100 From: ;tag=27317 To: ;tag=as071e36af Call-ID: 6994090221311298767-1168941511@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14350 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:32] DEBUG[18358]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:32] DEBUG[18358]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:32] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:32] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:32] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5d83a3d4 To: ;tag=t1168941512-co1395 From: "937654922" ;tag=as2aed489a Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1395-CPO02817 Content-Type: application/sdp Content-Length: 228 v=0 o=- 652719331 652719331 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 41144 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5d83a3d4 (64) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941512-co1395 (58) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "937654922" ;tag=as2aed489a (62) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 (55) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1395-CPO02817 (61) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 652719331 652719331 IN IP4 216.226.69.244 (45) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 41144 RTP/AVP 18 101 (28) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0d006a9841c11388370adc3 b5e2c408b@217.113.64.11' Request 102: Found [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:41144 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f02cd8 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:41144 [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:32] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:32] DEBUG[18358]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:32] DEBUG[18358]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 -- SIP/216.226.69.244-09f02cd8 is making progress passing it to SIP/5060-09d967f8 [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '11afa1dd5f8e2c414da3087e4ee762a5@217.113.64.11' [Jan 16 10:58:33] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 11afa1dd5f8e2c414da3087e4ee762a5@217.113.64.11 Really destroying SIP dialog '11afa1dd5f8e2c414da3087e4ee762a5@217.113.64.11' Method: INVITE [Jan 16 10:58:33] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:33] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:33] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:33] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205667397@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298771-1168941513@172.16.100.100 From: ;tag=28226 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3520717 Contact: sip:974243730@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3771 3771 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 62452 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205667397@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298771-1168941513@172.16.100.100 (54) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=28226 (62) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3 520717 (85) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:974243730@172.16.100.100:5060;user=phone (53) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3771 3771 IN IP4 10.57.3.22 (40) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 62452 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298771-1168941513@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298771-1168941513@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:62452 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.22:62452 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205667397 in default (domain 10.100.20.11) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:974243730@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a02bbd8: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3520717;received=172.16.100.100 From: ;tag=28226 To: Call-ID: 6994090221311298771-1168941513@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:34] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a02bbd8 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:34] DEBUG[18359]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432205667397@default:1] Dial("SIP/5060-0a02bbd8", "SIP/2205667397@216.226.69.244||t") in new stack [Jan 16 10:58:34] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:34] DEBUG[18359]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432205667397-1. [Jan 16 10:58:34] DEBUG[18359]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:34] DEBUG[18359]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:34] DEBUG[18359]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:2804 sip_call: Outgoing Call for 2205667397 [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 13110 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205667397@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK67f60129;rport (64) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 2: From: "974243730" ;tag=as29956ef5 (62) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 (55) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:34 GMT (35) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: m=audio 13110 RTP/AVP 18 (24) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205667397@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK67f60129;rport From: "974243730" ;tag=as29956ef5 To: Contact: Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 13110 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #173 -- Called 2205667397@216.226.69.244 [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:34] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:34] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5087d470 To: ;tag=t1168941490-co1280 From: "972858126" ;tag=as09098cc7 Call-ID: 59b1a4c46fcb00662e934b306ad9a8fd@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1280-CPO02399 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1718824694 1718824694 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43164 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5087d470 (64) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941490-co1280 (58) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972858126" ;tag=as09098cc7 (62) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 59b1a4c46fcb00662e934b306ad9a8fd@217.113.64.11 (55) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1280-CPO02399 (61) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1718824694 1718824694 IN IP4 216.226.69.244 (47) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43164 RTP/AVP 18 101 (28) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '59b1a4c46fcb00662e934b306ad9a8fd@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43164 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09ec89e8 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43164 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 59b1a4c46fcb00662e934b306ad9a8fd@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2205666018@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK47735575;rport From: "972858126" ;tag=as09098cc7 To: ;tag=t1168941490-co1280 Contact: Call-ID: 59b1a4c46fcb00662e934b306ad9a8fd@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:34] DEBUG[18349]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09ec89e8 -- SIP/216.226.69.244-09ec89e8 answered SIP/5060-09ec6bd0 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:34] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:34] DEBUG[18349]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09ec6bd0 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09ec6bd0 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:34] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19504 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c95-ac106464-3520681;received=172.16.100.100 From: ;tag=1393 To: ;tag=as6b7c6a30 Call-ID: 6994090221311298709-1168941490@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19504 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:34] DEBUG[18349]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #175 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205666018@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298709-1168941490@172.16.100.100 From: ;tag=1393 To: ;tag=as6b7c6a30 Seq: 1 ACKCLI> Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c95-ac106464-3520681 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205666018@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298709-1168941490@172.16.100.100 (54) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1393 (61) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as6b7c6a30 (67) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c95-ac106464-3 520681 (85) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #175 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298709-1168941490@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:34] DEBUG[18349]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1308eed8 To: ;tag=t1168941493-co1291 From: "621910128" ;tag=as1f76c99c Call-ID: 386ac3f2467cda4712c9c5a443a37062@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1291-CPO02443 Content-Type: application/sdp Content-Length: 228 v=0 o=- 649063686 649063686 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43216 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1308eed8 (64) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941493-co1291 (58) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "621910128" ;tag=as1f76c99c (62) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 386ac3f2467cda4712c9c5a443a37062@217.113.64.11 (55) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1291-CPO02443 (61) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 649063686 649063686 IN IP4 216.226.69.244 (45) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43216 RTP/AVP 18 101 (28) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '386ac3f2467cda4712c9c5a443a37062@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43216 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09ecd160 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43216 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 386ac3f2467cda4712c9c5a443a37062@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2208100530@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2220a95d;rport From: "621910128" ;tag=as1f76c99c To: ;tag=t1168941493-co1291 Contact: Call-ID: 386ac3f2467cda4712c9c5a443a37062@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms ax-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:34] DEBUG[18350]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09ecd160 -- SIP/216.226.69.244-09ecd160 answered SIP/5060-09f24658 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:34] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:34] DEBUG[18350]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09f24658 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09f24658 [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:34] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:34] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11742 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ca1-ac106464-3520687;received=172.16.100.100 From: ;tag=19140 To: ;tag=as3107366b Call-ID: 6994090221311298721-1168941493@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11742 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:34] DEBUG[18350]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #176 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432208100530@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298721-1168941493@172.16.100.100 From: ;tag=19140 To: ;tag=as3107366b CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ca1-ac106464-3520687 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432208100530@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298721-1168941493@172.16.100.100 (54) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=19140 (62) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as3107366b (67) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ca1-ac106464-3 520687 (85) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #176 [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298721-1168941493@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:34] DEBUG[18350]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK67f60129 To: From: "974243730" ;tag=as29956ef5 Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1416-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK67f60129 (64) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "974243730" ;tag=as29956ef5 (62) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 (55) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1416-CPO00000 (58) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #173 - INVITE (got response) [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '16e903bb7c0f03c0374daba b44d9dfc8@217.113.64.11' Request 102: Found [Jan 16 10:58:34] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:34] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 16138 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:34] DEBUG[18359]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3520717;received=172.16.100.100 From: ;tag=28226 To: ;tag=as66c721f7 Call-ID: 6994090221311298771-1168941513@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 16138 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:34] DEBUG[18359]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:34] DEBUG[18359]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:34] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:34] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK67f60129 To: ;tag=t1168941514-co1416 From: "974243730" ;tag=as29956ef5 Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1416-CPO02897 Content-Type: application/sdp Content-Length: 228 v=0 o=- 892334078 892334078 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 42488 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK67f60129 (64) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941514-co1416 (58) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "974243730" ;tag=as29956ef5 (62) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 (55) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1416-CPO02897 (61) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 892334078 892334078 IN IP4 216.226.69.244 (45) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42488 RTP/AVP 18 101 (28) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '16e903bb7c0f03c0374daba b44d9dfc8@217.113.64.11' Request 102: Found [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:42488 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f47100 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:42488 [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:35] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-09f47100 is making progress passing it to SIP/5060-0a02bbd8 [Jan 16 10:58:35] DEBUG[18359]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:35] DEBUG[18359]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18352]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:35] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK0887654f To: ;tag=t1168941500-co1337 From: "243851457441" ;tag=as702f17db Call-ID: 71f3d99662442a45574a9abb4681c4b3@217.113.64.11 CSeq: 102 INVITE Reason: Q.850 ;cause=28 ;text="Unknown" Contact: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1337-CPO02595 Content-Length: 0 <-------------> [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 484 Address Incomplete (30) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK0887654f (64) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941500-co1337 (57) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "243851457441" ;tag=as702f17db (68) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 71f3d99662442a45574a9abb4681c4b3@217.113.64.11 (55) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850 ;cause=28 ;text="Unknown" (39) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: (44) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1337-CPO02595 (58) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 0 (17) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '71f3d99662442a45574a9abb4681c4b3@217.113.64.11' of Request 102: M atch Not Found -- Got SIP response 484 "Address Incomplete" back from 216.226.69.244 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:220985483@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0887654f;rport From: "243851457441" ;tag=as702f17db To: ;tag=t1168941500-co1337 Contact: Call-ID: 71f3d99662442a45574a9abb4681c4b3@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 71f3d99662442a45574a9abb4681c4b3@217.113.64.11 [Jan 16 10:58:36] DEBUG[18352]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-09ddd560' [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-09ddd560, SIP callid 71f3d99662442a45574a9abb4681c4b3@217.113 .64.11) [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:3278 sip_hangup: update_call_counter(220985483) - decrement call limit counter on hangup [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:36] DEBUG[18352]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09ddd560 == Everyone is busy/congested at this time (1:0/0/1) [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18352]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:36] DEBUG[18352]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CHANUNAVAIL. [Jan 16 10:58:36] DEBUG[18352]: pbx.c:1768 pbx_extension_helper: Launching 'Goto' -- Executing [043220985483@default:2] Goto("SIP/5060-09e99548", "s-CHANUNAVAIL|1") in new stack [Jan 16 10:58:36] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 -- Goto (default,s-CHANUNAVAIL,1) [Jan 16 10:58:36] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:36] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:36] DEBUG[18352]: pbx.c:1768 pbx_extension_helper: Launching 'Hangup' -- Executing [s-CHANUNAVAIL@default:1] Hangup("SIP/5060-09e99548", "") in new stack [Jan 16 10:58:36] DEBUG[18352]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,s-CHANUNAVAIL,1) exited non-zero on 'SIP/5060-09e99548' == Spawn extension (default, s-CHANUNAVAIL, 1) exited non-zero on 'SIP/5060-09e99548' [Jan 16 10:58:36] DEBUG[18352]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:36] DEBUG[18352]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09e99548' [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09e99548, SIP callid 6994090221311298743-1168941499@172.16.100.100) [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311298743-1168941499@172.16.100.100' in 32000 ms (Method: INVITE) localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 484 Address incomplete Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb7-ac106464-3520697;received=172.16.100.100 From: ;tag=21866 To: ;tag=as714779a1 Call-ID: 6994090221311298743-1168941499@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:58:36] DEBUG[18352]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #180 [Jan 16 10:58:36] DEBUG[18352]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09e99548 [Jan 16 10:58:36] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:36] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:36] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) <--- SIP read from 172.16.100.100:5060 ---> ACK sip:043220985483@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298743-1168941499@172.16.100.100 From: ;tag=21866 To: ;tag=as714779a1 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb7-ac106464-3520697 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:043220985483@10.100.20.11:5060;user=phone SIP/2.0 (57) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298743-1168941499@172.16.100.100 (54) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=21866 (65) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as714779a1 (66) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cb7-ac106464-3 520697 (85) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #180 [Jan 16 10:58:36] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298743-1168941499@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '71f3d99662442a45574a9abb4681c4b3@217.113.64.11' Method: INVITE [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:36] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: OPTIONS sip:2007@10.100.20.190 SIP/2.0 (38) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1c97c393;rport (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: "asterisk" ;tag=as4e7d62f4 (59) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (28) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Contact: (36) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 7a1c05f254a44fa76ec8c26d78220fa5@10.100.20.11 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:37 GMT (35) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 10.100.20.190:5060: OPTIONS sip:2007@10.100.20.190 SIP/2.0 Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1c97c393;rport From: "asterisk" ;tag=as4e7d62f4 To: Contact: Call-ID: 7a1c05f254a44fa76ec8c26d78220fa5@10.100.20.11 CSeq: 102 OPTIONS User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #181 [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 10.100.20.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1c97c393;rport From: "asterisk" ;tag=as4e7d62f4 To: ;tag=91D126BA-2FFB531 CSeq: 102 OPTIONS Call-ID: 7a1c05f254a44fa76ec8c26d78220fa5@10.100.20.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Length: 0 <-------------> [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1c97c393;rport (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: "asterisk" ;tag=as4e7d62f4 (59) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=91D126BA-2FFB531 (49) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 7a1c05f254a44fa76ec8c26d78220fa5@10.100.20.11 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (33) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRA CK, UPDATE, REFER (96) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 (49) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 0 (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #181 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '7a1c05f254a44fa76ec8c26d78220fa5@10.100.20.11' of Request 102: Ma tch Not Found Really destroying SIP dialog '7a1c05f254a44fa76ec8c26d78220fa5@10.100.20.11' Method: OPTIONS [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: OPTIONS sip:2006@10.100.20.192 SIP/2.0 (38) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK4752a1bd;rport (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: "asterisk" ;tag=as5b6e9f25 (59) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (28) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Contact: (36) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 55957c8f7e01c2ca78a0bb1c0a83bd49@10.100.20.11 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:37 GMT (35) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 10.100.20.192:5060: OPTIONS sip:2006@10.100.20.192 SIP/2.0 Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK4752a1bd;rport From: "asterisk" ;tag=as5b6e9f25 To: Contact: Call-ID: 55957c8f7e01c2ca78a0bb1c0a83bd49@10.100.20.11 CSeq: 102 OPTIONS User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #184 [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 10.100.20.192:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK4752a1bd;rport From: "asterisk" ;tag=as5b6e9f25 To: ;tag=E9F91D69-96725058 CSeq: 102 OPTIONS Call-ID: 55957c8f7e01c2ca78a0bb1c0a83bd49@10.100.20.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Length: 0 <-------------> [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK4752a1bd;rport (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: "asterisk" ;tag=as5b6e9f25 (59) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=E9F91D69-96725058 (50) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 55957c8f7e01c2ca78a0bb1c0a83bd49@10.100.20.11 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (33) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRA CK, UPDATE, REFER (96) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 (49) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 0 (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #184 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '55957c8f7e01c2ca78a0bb1c0a83bd49@10.100.20.11' of Request 102: Ma tch Not Found Really destroying SIP dialog '55957c8f7e01c2ca78a0bb1c0a83bd49@10.100.20.11' Method: OPTIONS [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: OPTIONS sip:2004@10.100.20.193 SIP/2.0 (38) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1528bfd2;rport (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: "asterisk" ;tag=as2c02f562 (59) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (28) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Contact: (36) [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 05f19a94765134c775aa90565f53730d@10.100.20.11 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:37 GMT (35) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 10.100.20.193:5060: OPTIONS sip:2004@10.100.20.193 SIP/2.0 Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1528bfd2;rport From: "asterisk" ;tag=as2c02f562 To: Contact: Call-ID: 05f19a94765134c775aa90565f53730d@10.100.20.11 CSeq: 102 OPTIONS User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #187 [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:37] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:37] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 10.100.20.193:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1528bfd2;rport From: "asterisk" ;tag=as2c02f562 To: ;tag=FCE93A98-78BA0EA1 CSeq: 102 OPTIONS Call-ID: 05f19a94765134c775aa90565f53730d@10.100.20.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Length: 0 <-------------> [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK1528bfd2;rport (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: "asterisk" ;tag=as2c02f562 (59) [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=FCE93A98-78BA0EA1 (50) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 05f19a94765134c775aa90565f53730d@10.100.20.11 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (33) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRA CK, UPDATE, REFER (96) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 (49) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 0 (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #187 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '05f19a94765134c775aa90565f53730d@10.100.20.11' of Request 102: Ma tch Not Found Really destroying SIP dialog '05f19a94765134c775aa90565f53730d@10.100.20.11' Method: OPTIONS [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK09e0fb00 To: ;tag=t1168941503-co1352 From: "34972244038" ;tag=as50e60bc0 Call-ID: 314d7a4a51948502362647460e8b8c33@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1352-CPO02650 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1214092970 1214092970 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 51016 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK09e0fb00 (64) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941503-co1352 (58) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "34972244038" ;tag=as50e60bc0 (66) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 314d7a4a51948502362647460e8b8c33@217.113.64.11 (55) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1352-CPO02650 (61) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1214092970 1214092970 IN IP4 216.226.69.244 (47) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 51016 RTP/AVP 18 101 (28) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '314d7a4a51948502362647460e8b8c33@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:51016 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0a081c20 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.19.106.196:51016 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 314d7a4a51948502362647460e8b8c33@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2209897131@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0128c78c;rport From: "34972244038" ;tag=as50e60bc0 To: ;tag=t1168941503-co1352 Contact: Call-ID: 314d7a4a51948502362647460e8b8c33@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:37] DEBUG[18355]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0a081c20 -- SIP/216.226.69.244-0a081c20 answered SIP/5060-0a06fb20 [Jan 16 10:58:37] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:37] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:37] DEBUG[18355]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a06fb20 [Jan 16 10:58:37] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-0a06fb20 [Jan 16 10:58:37] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:37] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:37] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14228 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc1-ac106464-3520702;received=172.16.100.100 From: ;tag=6844 To: ;tag=as5a91d2e6 Call-ID: 6994090221311298753-1168941502@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14228 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:37] DEBUG[18355]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #190 [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432209897131@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298753-1168941502@172.16.100.100 From: ;tag=6844 To: ;tag=as5a91d2e6 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc1-ac106464-3520702 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432209897131@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298753-1168941502@172.16.100.100 (54) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=6844 (63) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as5a91d2e6 (67) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc1-ac106464-3 520702 (85) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #190 [Jan 16 10:58:37] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298753-1168941502@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18355]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:37] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:38] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:39] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '6994090221311298699-1168941485@172.16.100.100' [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 6994090221311298699-1168941485@172.16.100.100 Really destroying SIP dialog '6994090221311298699-1168941485@172.16.100.100' Method: ACK [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18351]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432208901919@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298733-1168941498@172.16.100.100 From: ;tag=5027 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cad-ac106464-3520693 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432208901919@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298733-1168941498@172.16.100.100 (54) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=5027 (61) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cad-ac106464-3 520693 (85) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298733-1168941498@172.16.100.100 localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cad-ac106464-3520693;received=172.16.100.100 From: ;tag=5027 To: ;tag=as008fde26 Call-ID: 6994090221311298733-1168941498@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #191 localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cad-ac106464-3520693;received=172.16.100.100 From: ;tag=5027 To: ;tag=as008fde26 Call-ID: 6994090221311298733-1168941498@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:39] DEBUG[18351]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:39] DEBUG[18351]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-09dc5478' [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-09dc5478, SIP callid 7185817751d170532b4bce912c8cd142@217.113 .64.11) [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:3278 sip_hangup: update_call_counter(2208901919) - decrement call limit counter on hangup [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '7185817751d170532b4bce912c8cd142@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '7185817751d170532b4bce912c8cd142@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2208901919@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK271ae837;rport From: "972370597" ;tag=as23aaa2ae To: Call-ID: 7185817751d170532b4bce912c8cd142@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #193 Scheduling destruction of SIP dialog '7185817751d170532b4bce912c8cd142@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:39] DEBUG[18351]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09dc5478 [Jan 16 10:58:39] DEBUG[18351]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:39] DEBUG[18351]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432208901919,1) exited non-zero on 'SIP/5060-09d74158' == Spawn extension (default, 0432208901919, 1) exited non-zero on 'SIP/5060-09d74158' [Jan 16 10:58:39] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:39] DEBUG[18351]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:39] DEBUG[18351]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09d74158' [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09d74158, SIP callid 6994090221311298733-1168941498@172.16.100.100) [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:39] DEBUG[18351]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:58:39] DEBUG[18351]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d74158 [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:39] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432208901919@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298733-1168941498@172.16.100.100 From: ;tag=5027 To: ;tag=as008fde26 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cad-ac106464-3520693 Max-Forwards: 70 Content-Length: 0 localhost*CLI> <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432208901919@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298733-1168941498@172.16.100.100 (54) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=5027 (61) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as008fde26 (67) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cad-ac106464-3 520693 (85) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #191 [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298733-1168941498@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311298733-1168941498@172.16.100.100' Method: ACK [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5d83a3d4 To: ;tag=t1168941512-co1395 From: "937654922" ;tag=as2aed489a Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1395-CPO02817 Content-Type: application/sdp Content-Length: 228 v=0 o=- 652719331 652719331 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 41144 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5d83a3d4 (64) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941512-co1395 (58) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "937654922" ;tag=as2aed489a (62) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 (55) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1395-CPO02817 (61) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 652719331 652719331 IN IP4 216.226.69.244 (45) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 41144 RTP/AVP 18 101 (28) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '0d006a9841c11388370adc3b5e2c408b@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:41144 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f02cd8 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:41144 [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2205667787@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5c6fb3c8;rport From: "937654922" ;tag=as2aed489a To: ;tag=t1168941512-co1395 Contact: Call-ID: 0d006a9841c11388370adc3b5e2c408b@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:39] DEBUG[18358]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09f02cd8 -- SIP/216.226.69.244-09f02cd8 answered SIP/5060-09d967f8 [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:39] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:39] DEBUG[18358]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d967f8 [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09d967f8 [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:39] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:39] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14350 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3520714;received=172.16.100.100 From: ;tag=27317 To: ;tag=as071e36af Call-ID: 6994090221311298767-1168941511@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14350 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:39] DEBUG[18358]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #195 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205667787@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298767-1168941511@172.16.100.100 From: ;tag=27317 To: ;tag=as071e36af CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3520714 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205667787@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298767-1168941511@172.16.100.100 (54) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=27317 (62) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as071e36af (67) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccf-ac106464-3 520714 (85) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #195 [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298767-1168941511@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INFO sip:0432209897131@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298753-1168941502@172.16.100.100 From: ;tag=6844 To: ;tag=as5a91d2e6 CSeq: 2 INFO Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc1-ac106464-3520724 Supported: timer,100rel Max-Forwards: 70 Content-Length: 11 Content-Type: application/dtmf-relay Signal= # localhost*CLI> <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INFO sip:0432209897131@10.100.20.11:5060;user=phone SIP/2.0 (59) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298753-1168941502@172.16.100.100 (54) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=6844 (63) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as5a91d2e6 (67) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 2 INFO (12) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc1-ac106464-3 520724 (85) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 11 (18) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Type: application/dtmf-relay (36) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: Signal= # (9) --- (10 headers 1 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INFO (13) - Command in SIP INFO Receiving INFO! * DTMF-relay event received: # localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cc1-ac106464-3520724;received=172.16.100.100 From: ;tag=6844 To: ;tag=as5a91d2e6 Call-ID: 6994090221311298753-1168941502@172.16.100.100 CSeq: 2 INFO User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:39] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK271ae837 To: ;tag=t1168941498-co1320 From: "972370597" ;tag=as23aaa2ae Call-ID: 7185817751d170532b4bce912c8cd142@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1320-CPO02540 Content-Length: 0 <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK271ae837 (64) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941498-co1320 (58) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972370597" ;tag=as23aaa2ae (62) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 7185817751d170532b4bce912c8cd142@217.113.64.11 (55) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1320-CPO02540 (61) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #193 [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '7185817751d170532b4bce912c8cd142@217.113.64.11' of Request 102: M atch Not Found localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK271ae837 To: ;tag=t1168941498-co1320 From: "972370597" ;tag=as23aaa2ae Call-ID: 7185817751d170532b4bce912c8cd142@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1320-CPO02540 Content-Length: 0 <-------------> [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK271ae837 (64) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941498-co1320 (58) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972370597" ;tag=as23aaa2ae (62) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 7185817751d170532b4bce912c8cd142@217.113.64.11 (55) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1320-CPO02540 (61) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '7185817751d170532b4bce912c8cd142@217.113.64.11' of Request 102: M atch Found [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2208901919@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK271ae837;rport From: "972370597" ;tag=as23aaa2ae To: ;tag=t1168941498-co1320 Contact: Call-ID: 7185817751d170532b4bce912c8cd142@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:39] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18355]: chan_sip.c:5576 reqprep: Strict routing enforced for session 314d7a4a51948502362647460e8b8c33@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Reliably Transmitting (no NAT) to 216.226.69.244:5060: INFO sip:2209897131@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7e309f6c;rport From: "34972244038" ;tag=as50e60bc0 To: ;tag=t1168941503-co1352 Contact: Call-ID: 314d7a4a51948502362647460e8b8c33@217.113.64.11 CSeq: 103 INFO User-Agent: gatewaycomms Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=# Duration=250 --- [Jan 16 10:58:39] DEBUG[18355]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #196 [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:39] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298781-1168941520@172.16.100.100 From: ;tag=15022 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520725 Contact: sip:004769838857@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3267 3267 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 61348 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298781-1168941520@172.16.100.100 (54) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=15022 (65) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3 520725 (85) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:004769838857@172.16.100.100:5060;user=phone (56) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3267 3267 IN IP4 10.57.3.22 (40) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 61348 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298781-1168941520@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298781-1168941520@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:61348 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.22:61348 [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209825337 in default (domain 10.100.20.11) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:004769838857@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09d74158: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520725;received=172.16.100.100 From: ;tag=15022 To: Call-ID: 6994090221311298781-1168941520@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:40] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d74158 [Jan 16 10:58:40] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:40] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:40] DEBUG[18360]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432209825337@default:1] Dial("SIP/5060-09d74158", "SIP/2209825337@216.226.69.244||t") in new stack [Jan 16 10:58:40] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:40] DEBUG[18360]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432209825337-1. [Jan 16 10:58:40] DEBUG[18360]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:40] DEBUG[18360]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:40] DEBUG[18360]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:2804 sip_call: Outgoing Call for 2209825337 [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 16770 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209825337@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK23e28bc4;rport (64) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 2: From: "004769838857" ;tag=as1f5067ce (68) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 (55) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:40 GMT (35) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: m=audio 16770 RTP/AVP 18 (24) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209825337@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK23e28bc4;rport From: "004769838857" ;tag=as1f5067ce To: Contact: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16770 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #197 -- Called 2209825337@216.226.69.244 [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7e309f6c To: ;tag=t1168941503-co1352 From: "34972244038" ;tag=as50e60bc0 Call-ID: 314d7a4a51948502362647460e8b8c33@217.113.64.11 CSeq: 103 INFO User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1352-CPO02650 Content-Length: 0 <-------------> [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7e309f6c (64) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941503-co1352 (58) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "34972244038" ;tag=as50e60bc0 (66) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 314d7a4a51948502362647460e8b8c33@217.113.64.11 (55) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 103 INFO (14) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1352-CPO02650 (61) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #196 [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '314d7a4a51948502362647460e8b8c33@217.113.64.11' of Request 103: M atch Not Found [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK23e28bc4 To: From: "004769838857" ;tag=as1f5067ce Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1455-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK23e28bc4 (64) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "004769838857" ;tag=as1f5067ce (68) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 (55) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1455-CPO00000 (58) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #197 - INVITE (got response) [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '126a554e5a5c42445478483 7291c97de@217.113.64.11' Request 102: Found [Jan 16 10:58:40] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11656 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:40] DEBUG[18360]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520725;received=172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 Call-ID: 6994090221311298781-1168941520@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11656 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:40] DEBUG[18360]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:40] DEBUG[18360]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:40] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:41] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:41] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:42] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:42] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK23e28bc4 To: ;tag=t1168941520-co1455 From: "004769838857" ;tag=as1f5067ce Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 Content-Type: application/sdp Content-Length: 230 =0calhost*CLI> o=- 1271377745 1271377745 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 52212 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK23e28bc4 (64) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941520-co1455 (58) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "004769838857" ;tag=as1f5067ce (68) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 (55) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 (61) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1271377745 1271377745 IN IP4 216.226.69.244 (47) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 52212 RTP/AVP 18 101 (28) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '126a554e5a5c42445478483 7291c97de@217.113.64.11' Request 102: Found [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:52212 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09ddd560 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.19.106.196:52212 [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-09ddd560 is making progress passing it to SIP/5060-09d74158 [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18360]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:43] DEBUG[18360]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249915583309@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298789-1168941523@172.16.100.100 From: ;tag=1 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729 Contact: sip:2356652452@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 9877 9877 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 42496 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249915583309@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298789-1168941523@172.16.100.100 (54) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1 (59) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3 520729 (85) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356652452@172.16.100.100:5060;user=phone (54) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9877 9877 IN IP4 10.57.2.121 (41) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42496 RTP/AVP 18 101 13 (31) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298789-1168941523@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298789-1168941523@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:42496 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.121:42496 [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249915583309 in default (domain 10.100.20.11) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356652452@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:43] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09d8a738: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729;received=172.16.100.100 From: ;tag=1 To: Call-ID: 6994090221311298789-1168941523@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:43] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d8a738 [Jan 16 10:58:43] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:43] DEBUG[18361]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0264249915583309@default:1] Dial("SIP/5060-09d8a738", "SIP/249915583309@196.29.163.6||t") in new stack [Jan 16 10:58:43] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:43] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:43] DEBUG[18361]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0264249915583309-1. [Jan 16 10:58:43] DEBUG[18361]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:43] DEBUG[18361]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:43] DEBUG[18361]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:2804 sip_call: Outgoing Call for 249915583309 [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 11630 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249915583309@196.29.163.6 SIP/2.0 (44) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport (64) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 2: From: "2356652452" ;tag=as0034b53c (64) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 (55) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:43 GMT (35) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: m=audio 11630 RTP/AVP 18 (24) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249915583309@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport From: "2356652452" ;tag=as0034b53c To: ontact: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 11630 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:43] DEBUG[18361]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #201 -- Called 249915583309@196.29.163.6 [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:43] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356652452" ;tag=as0034b53c To: CSeq: 102 INVITE Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356652452" ;tag=as0034b53c (64) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 (55) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 (69) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #201 - INVITE (got response) [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '57c2d72d608b13b93f0a972 e6f4e6d56@217.113.64.11' Request 102: Found [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:44] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '712ed1c271c8073417d0b0bc36d3d568@217.113.64.11' [Jan 16 10:58:44] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 712ed1c271c8073417d0b0bc36d3d568@217.113.64.11 Really destroying SIP dialog '712ed1c271c8073417d0b0bc36d3d568@217.113.64.11' Method: INVITE [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:44] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:45] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:46] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:46] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:47] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:47] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '7f2ffdc0102566e744adf29d4f5f7930@217.113.64.11' [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 7f2ffdc0102566e744adf29d4f5f7930@217.113.64.11 Really destroying SIP dialog '7f2ffdc0102566e744adf29d4f5f7930@217.113.64.11' Method: INVITE [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209935770@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298795-1168941527@172.16.100.100 From: ;tag=1818 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3520731 Contact: sip:933122088@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3419 3419 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 20480 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209935770@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298795-1168941527@172.16.100.100 (54) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1818 (61) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3 520731 (85) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:933122088@172.16.100.100:5060;user=phone (53) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3419 3419 IN IP4 10.57.3.22 (40) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 20480 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298795-1168941527@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298795-1168941527@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:20480 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.22:20480 [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209935770 in default (domain 10.100.20.11) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:933122088@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:47] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09dc79f0: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3520731;received=172.16.100.100 From: ;tag=1818 To: Call-ID: 6994090221311298795-1168941527@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09dc79f0 [Jan 16 10:58:47] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:47] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:47] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:47] DEBUG[18362]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432209935770@default:1] Dial("SIP/5060-09dc79f0", "SIP/2209935770@216.226.69.244||t") in new stack [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:47] DEBUG[18362]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432209935770-1. [Jan 16 10:58:47] DEBUG[18362]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:47] DEBUG[18362]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:47] DEBUG[18362]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:2804 sip_call: Outgoing Call for 2209935770 [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 17830 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209935770@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3b9ac359;rport (64) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 2: From: "933122088" ;tag=as419958cb (62) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 (55) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:47 GMT (35) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: m=audio 17830 RTP/AVP 18 (24) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209935770@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3b9ac359;rport From: "933122088" ;tag=as419958cb To: Contact: Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 17830 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:47] DEBUG[18362]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #203 -- Called 2209935770@216.226.69.244 [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:47] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK67f60129 To: ;tag=t1168941514-co1416 From: "974243730" ;tag=as29956ef5 Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1416-CPO02897 Content-Type: application/sdp Content-Length: 228 v=0 o=- 892334078 892334078 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 42488 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK67f60129 (64) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941514-co1416 (58) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "974243730" ;tag=as29956ef5 (62) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 (55) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1416-CPO02897 (61) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 892334078 892334078 IN IP4 216.226.69.244 (45) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42488 RTP/AVP 18 101 (28) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:42488 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f47100 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:42488 [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2205667397@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK6a5a1109;rport From: "974243730" ;tag=as29956ef5 To: ;tag=t1168941514-co1416 Contact: Call-ID: 16e903bb7c0f03c0374dabab44d9dfc8@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:48] DEBUG[18359]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09f47100 -- SIP/216.226.69.244-09f47100 answered SIP/5060-0a02bbd8 [Jan 16 10:58:48] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:48] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:48] DEBUG[18359]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a02bbd8 [Jan 16 10:58:48] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-0a02bbd8 [Jan 16 10:58:48] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:48] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:48] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 16138 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3520717;received=172.16.100.100 From: ;tag=28226 To: ;tag=as66c721f7 Call-ID: 6994090221311298771-1168941513@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 16138 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:48] DEBUG[18359]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #205 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205667397@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298771-1168941513@172.16.100.100 From: ;tag=28226 To: ;tag=as66c721f7 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3520717 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205667397@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298771-1168941513@172.16.100.100 (54) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=28226 (62) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as66c721f7 (67) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cd3-ac106464-3 520717 (85) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #205 [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298771-1168941513@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:48] DEBUG[18359]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3b9ac359 To: From: "933122088" ;tag=as419958cb Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1492-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3b9ac359 (64) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "933122088" ;tag=as419958cb (62) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 (55) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1492-CPO00000 (58) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #203 - INVITE (got response) [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '79458a6c010104ae25e6ffb f3851550d@217.113.64.11' Request 102: Found [Jan 16 10:58:48] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:48] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:48] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:48] DEBUG[18362]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:48] DEBUG[18362]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:48] DEBUG[18362]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14286 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:48] DEBUG[18362]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:48] DEBUG[18362]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3520731;received=172.16.100.100 From: ;tag=1818 To: ;tag=as225a1882 Call-ID: 6994090221311298795-1168941527@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14286 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:48] DEBUG[18362]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:48] DEBUG[18362]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:48] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432204484010@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298799-1168941529@172.16.100.100 From: ;tag=2727 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3520733 Contact: sip:972402839@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3522 3522 IN IP4 10.57.3.26 s=- c=IN IP4 10.57.3.26 t=0 0 m=audio 42260 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432204484010@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298799-1168941529@172.16.100.100 (54) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=2727 (61) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3 520733 (85) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:972402839@172.16.100.100:5060;user=phone (53) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3522 3522 IN IP4 10.57.3.26 (40) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.26 (19) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42260 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298799-1168941529@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298799-1168941529@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.26:42260 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.26:42260 [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432204484010 in default (domain 10.100.20.11) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:972402839@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09d94ff0: New call is still down.... Trying... localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3520733;received=172.16.100.100 From: ;tag=2727 To: Call-ID: 6994090221311298799-1168941529@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:49] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d94ff0 [Jan 16 10:58:49] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:49] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:49] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:49] DEBUG[18363]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432204484010@default:1] Dial("SIP/5060-09d94ff0", "SIP/2204484010@216.226.69.244||t") in new stack [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:49] DEBUG[18363]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432204484010-1. [Jan 16 10:58:49] DEBUG[18363]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:49] DEBUG[18363]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:49] DEBUG[18363]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:2804 sip_call: Outgoing Call for 2204484010 [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 16232 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2204484010@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK04843c9a;rport (64) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 2: From: "972402839" ;tag=as2f3d0882 (62) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 (55) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:49 GMT (35) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: m=audio 16232 RTP/AVP 18 (24) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2204484010@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK04843c9a;rport From: "972402839" ;tag=as2f3d0882 To: Contact: Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16232 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #207 -- Called 2204484010@216.226.69.244 [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298797-1168941528@172.16.100.100 From: ;tag=18656 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ced-ac106464-3520734 Contact: sip:2356756282@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 9882 9882 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 56864 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298797-1168941528@172.16.100.100 (54) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=18656 (63) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ced-ac106464-3 520734 (85) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356756282@172.16.100.100:5060;user=phone (54) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9882 9882 IN IP4 10.57.2.121 (41) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 56864 RTP/AVP 18 101 13 (31) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298797-1168941528@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298797-1168941528@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:56864 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.121:56864 [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249734822305 in default (domain 10.100.20.11) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356756282@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a117120: New call is still down.... Trying... [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ced-ac106464-3520734;received=172.16.100.100 From: ;tag=18656 To: Call-ID: 6994090221311298797-1168941528@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a117120 [Jan 16 10:58:49] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:49] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:49] DEBUG[18364]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0264249734822305@default:1] Dial("SIP/5060-0a117120", "SIP/249734822305@196.29.163.6||t") in new stack [Jan 16 10:58:49] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:49] DEBUG[18364]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0264249734822305-1. [Jan 16 10:58:49] DEBUG[18364]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:49] DEBUG[18364]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:49] DEBUG[18364]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:2804 sip_call: Outgoing Call for 249734822305 [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 15814 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249734822305@196.29.163.6 SIP/2.0 (44) [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport (64) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 2: From: "2356756282" ;tag=as3a92afc0 (64) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 (55) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:49 GMT (35) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: m=audio 15814 RTP/AVP 18 (24) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport From: "2356756282" ;tag=as3a92afc0 To: Contact: Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 15814 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:49] DEBUG[18364]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #209 -- Called 249734822305@196.29.163.6 [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356756282" ;tag=as3a92afc0 To: CSeq: 102 INVITE Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356756282" ;tag=as3a92afc0 (64) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 (55) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport=5060 (69) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #209 - INVITE (got response) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4c01539736f60f4f303b99b b313c9e6a@217.113.64.11' Request 102: Found [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:49] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK04843c9a To: From: "972402839" ;tag=as2f3d0882 Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1503-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK04843c9a (64) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972402839" ;tag=as2f3d0882 (62) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 (55) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1503-CPO00000 (58) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #207 - INVITE (got response) [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3b86aa9e1a51c7a672149ed 3529fb28c@217.113.64.11' Request 102: Found [Jan 16 10:58:49] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14642 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:49] DEBUG[18363]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3520733;received=172.16.100.100 From: ;tag=2727 To: ;tag=as3756653a Call-ID: 6994090221311298799-1168941529@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14642 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:49] DEBUG[18363]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:49] DEBUG[18363]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:49] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 404 Not Found From: "2356756282" ;tag=as3a92afc0 To: ;tag=2dea8178 CSeq: 102 INVITE Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport=5060 Reason: Q.850;cause=1;text="Unallocated number" Content-Length: 0 <-------------> [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 404 Not Found (21) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356756282" ;tag=as3a92afc0 (64) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=2dea8178 (48) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 (55) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport=5060 (69) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850;cause=1;text="Unallocated number" (47) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '4c01539736f60f4f303b99bb313c9e6a@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 404 to standard invite Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK4f1b0917;rport From: "2356756282" ;tag=as3a92afc0 To: ;tag=2dea8178 Contact: Call-ID: 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 4c01539736f60f4f303b99bb313c9e6a@217.113.64.11 -- SIP/196.29.163.6-0a128690 is circuit-busy [Jan 16 10:58:50] DEBUG[18364]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-0a128690' [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-0a128690, SIP callid 4c01539736f60f4f303b99bb313c9e6a@217.113.6 4.11) [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:3278 sip_hangup: update_call_counter(249734822305) - decrement call limit counter on hangup [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:50] DEBUG[18364]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-0a128690 == Everyone is busy/congested at this time (1:0/1/0) [Jan 16 10:58:50] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 16 10:58:50] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 16 10:58:50] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) [Jan 16 10:58:50] DEBUG[18364]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:50] DEBUG[18364]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. [Jan 16 10:58:50] DEBUG[18364]: pbx.c:1768 pbx_extension_helper: Launching 'Goto' -- Executing [0264249734822305@default:2] Goto("SIP/5060-0a117120", "s-CONGESTION|1") in new stack -- Goto (default,s-CONGESTION,1) [Jan 16 10:58:50] DEBUG[18364]: pbx.c:1768 pbx_extension_helper: Launching 'Hangup' -- Executing [s-CONGESTION@default:1] Hangup("SIP/5060-0a117120", "") in new stack [Jan 16 10:58:50] DEBUG[18364]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,s-CONGESTION,1) exited non-zero on 'SIP/5060-0a117120' == Spawn extension (default, s-CONGESTION, 1) exited non-zero on 'SIP/5060-0a117120' [Jan 16 10:58:50] DEBUG[18364]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:50] DEBUG[18364]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-0a117120' [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-0a117120, SIP callid 6994090221311298797-1168941528@172.16.100.100) [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311298797-1168941528@172.16.100.100' in 32000 ms (Method: INVITE) localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ced-ac106464-3520734;received=172.16.100.100 From: ;tag=18656 To: ;tag=as64c73940 Call-ID: 6994090221311298797-1168941528@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:58:50] DEBUG[18364]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #213 [Jan 16 10:58:50] DEBUG[18364]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a117120 [Jan 16 10:58:50] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:50] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:50] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298797-1168941528@172.16.100.100 From: ;tag=18656 To: ;tag=as64c73940 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ced-ac106464-3520734 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298797-1168941528@172.16.100.100 (54) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=18656 (63) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as64c73940 (70) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ced-ac106464-3 520734 (85) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #213 [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298797-1168941528@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '4c01539736f60f4f303b99bb313c9e6a@217.113.64.11' Method: INVITE [Jan 16 10:58:50] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK04843c9a To: ;tag=t1168941529-co1503 From: "972402839" ;tag=as2f3d0882 Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1503-CPO00225 Content-Type: application/sdp Content-Length: 226 v=0 o=- 48898701 48898701 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 42620 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK04843c9a (64) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941529-co1503 (58) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972402839" ;tag=as2f3d0882 (62) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 (55) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1503-CPO00225 (61) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 226 (19) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 48898701 48898701 IN IP4 216.226.69.244 (43) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42620 RTP/AVP 18 101 (28) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3b86aa9e1a51c7a672149ed 3529fb28c@217.113.64.11' Request 102: Found [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:42620 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f51aa0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:42620 [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-09f51aa0 is making progress passing it to SIP/5060-09d94ff0 [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18363]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:50] DEBUG[18363]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3b9ac359 To: ;tag=t1168941528-co1492 From: "933122088" ;tag=as419958cb Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1492-CPO00194 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1205031023 1205031023 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 53156 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3b9ac359 (64) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941528-co1492 (58) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "933122088" ;tag=as419958cb (62) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 (55) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1492-CPO00194 (61) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1205031023 1205031023 IN IP4 216.226.69.244 (47) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 53156 RTP/AVP 18 101 (28) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '79458a6c010104ae25e6ffb f3851550d@217.113.64.11' Request 102: Found [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:53156 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f50520 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.19.106.196:53156 [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:50] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-09f50520 is making progress passing it to SIP/5060-09dc79f0 [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18362]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:50] DEBUG[18362]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:50] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249121063786@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298813-1168941530@172.16.100.100 From: ;tag=19565 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cfd-ac106464-3520739 Contact: sip:2356408270@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 9886 9886 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 60680 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249121063786@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298813-1168941530@172.16.100.100 (54) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=19565 (63) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cfd-ac106464-3 520739 (85) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356408270@172.16.100.100:5060;user=phone (54) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9886 9886 IN IP4 10.57.2.121 (41) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 60680 RTP/AVP 18 101 13 (31) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298813-1168941530@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298813-1168941530@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:60680 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.121:60680 [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249121063786 in default (domain 10.100.20.11) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356408270@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a1304f0: New call is still down.... Trying... localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cfd-ac106464-3520739;received=172.16.100.100 From: ;tag=19565 To: Call-ID: 6994090221311298813-1168941530@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:51] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a1304f0 [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:51] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:51] DEBUG[18365]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0264249121063786@default:1] Dial("SIP/5060-0a1304f0", "SIP/249121063786@196.29.163.6||t") in new stack [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:51] DEBUG[18365]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0264249121063786-1. [Jan 16 10:58:51] DEBUG[18365]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:51] DEBUG[18365]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:51] DEBUG[18365]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:2804 sip_call: Outgoing Call for 249121063786 [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 18066 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249121063786@196.29.163.6 SIP/2.0 (44) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK131f9f97;rport (64) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 2: From: "2356408270" ;tag=as1376ce66 (64) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 217bc3c0139062936630151f77da5ed9@217.113.64.11 (55) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:51 GMT (35) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: m=audio 18066 RTP/AVP 18 (24) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249121063786@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK131f9f97;rport From: "2356408270" ;tag=as1376ce66 To: Contact: Call-ID: 217bc3c0139062936630151f77da5ed9@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 18066 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:51] DEBUG[18365]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #216 -- Called 249121063786@196.29.163.6 [Jan 16 10:58:51] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:51] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356408270" ;tag=as1376ce66 To: CSeq: 102 INVITE Call-ID: 217bc3c0139062936630151f77da5ed9@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK131f9f97;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356408270" ;tag=as1376ce66 (64) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 217bc3c0139062936630151f77da5ed9@217.113.64.11 (55) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK131f9f97;rport=5060 (69) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #216 - INVITE (got response) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '217bc3c0139062936630151 f77da5ed9@217.113.64.11' Request 102: Found [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18333]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> BYE sip:0432209905473@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298625-1168941460@172.16.100.100 From: ;tag=20532 To: ;tag=as17142e87 CSeq: 2 BYE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c41-ac106464-3520742 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: BYE sip:0432209905473@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298625-1168941460@172.16.100.100 (54) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=20532 (64) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as17142e87 (67) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 2 BYE (11) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c41-ac106464-3 520742 (85) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298625-1168941460@172.16.100.100 [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:14079 handle_request_bye: Received bye, issuing owner hangup localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c41-ac106464-3520742;received=172.16.100.100 From: ;tag=20532 To: ;tag=as17142e87 Call-ID: 6994090221311298625-1168941460@172.16.100.100 CSeq: 2 BYE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:51] DEBUG[18333]: channel.c:3684 ast_generic_bridge: Didn't get a frame from channel: SIP/5060-09da0c70 [Jan 16 10:58:51] DEBUG[18333]: channel.c:3988 ast_channel_bridge: Bridge stops bridging channels SIP/5060-09da0c70 and SIP/216.226.69.244-09db30f0 [Jan 16 10:58:51] DEBUG[18333]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-09db30f0' [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-09db30f0, SIP callid 236e417e21635cf709fef7826f9b4722@217.113 .64.11) [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:3278 sip_hangup: update_call_counter(2209905473) - decrement call limit counter on hangup [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call Scheduling destruction of SIP dialog '236e417e21635cf709fef7826f9b4722@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:5576 reqprep: Strict routing enforced for session 236e417e21635cf709fef7826f9b4722@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Reliably Transmitting (no NAT) to 216.226.69.244:5060: BYE sip:2209905473@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK29fccab5;rport From: "23276647859" ;tag=as060ba4dd To: ;tag=t1168941460-co1126 Call-ID: 236e417e21635cf709fef7826f9b4722@217.113.64.11 CSeq: 103 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #219 [Jan 16 10:58:51] DEBUG[18333]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09db30f0 [Jan 16 10:58:51] DEBUG[18333]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:51] DEBUG[18333]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 16 10:58:51] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:51] DEBUG[18333]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432209905473,1) exited non-zero on 'SIP/5060-09da0c70' == Spawn extension (default, 0432209905473, 1) exited non-zero on 'SIP/5060-09da0c70' [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:51] DEBUG[18333]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:51] DEBUG[18333]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09da0c70' [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09da0c70, SIP callid 6994090221311298625-1168941460@172.16.100.100) [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:51] DEBUG[18333]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:51] DEBUG[18333]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09da0c70 [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:51] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:51] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed Really destroying SIP dialog '6994090221311298625-1168941460@172.16.100.100' Method: BYE [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18347]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432204489001@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298701-1168941487@172.16.100.100 From: ;tag=16414 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c8d-ac106464-3520679 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432204489001@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298701-1168941487@172.16.100.100 (54) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=16414 (63) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c8d-ac106464-3 520679 (85) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298701-1168941487@172.16.100.100 localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c8d-ac106464-3520679;received=172.16.100.100 From: ;tag=16414 To: ;tag=as2609d495 Call-ID: 6994090221311298701-1168941487@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #220 localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c8d-ac106464-3520679;received=172.16.100.100 From: ;tag=16414 To: ;tag=as2609d495 Call-ID: 6994090221311298701-1168941487@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:51] DEBUG[18347]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:51] DEBUG[18347]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-09f8e4e8' [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-09f8e4e8, SIP callid 4e1f2e773ca0de775d54ef0957ee6eed@217.113 .64.11) [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:3278 sip_hangup: update_call_counter(2204489001) - decrement call limit counter on hangup [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2204489001@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1171468a;rport From: "2425323347" ;tag=as6521d55e To: Call-ID: 4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #222 Scheduling destruction of SIP dialog '4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:51] DEBUG[18347]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09f8e4e8 [Jan 16 10:58:51] DEBUG[18347]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:51] DEBUG[18347]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432204489001,1) exited non-zero on 'SIP/5060-09f35b90' == Spawn extension (default, 0432204489001, 1) exited non-zero on 'SIP/5060-09f35b90' [Jan 16 10:58:51] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:51] DEBUG[18347]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:51] DEBUG[18347]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09f35b90' [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09f35b90, SIP callid 6994090221311298701-1168941487@172.16.100.100) [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:51] DEBUG[18347]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:58:51] DEBUG[18347]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09f35b90 [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:51] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:51] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432204489001@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298701-1168941487@172.16.100.100 From: ;tag=16414 To: ;tag=as2609d495 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c8d-ac106464-3520679 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432204489001@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298701-1168941487@172.16.100.100 (54) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=16414 (63) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as2609d495 (67) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0c8d-ac106464-3 520679 (85) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #220 [Jan 16 10:58:51] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298701-1168941487@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311298701-1168941487@172.16.100.100' Method: ACK [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:51] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK29fccab5 To: ;tag=t1168941460-co1126 From: "23276647859" ;tag=as060ba4dd Call-ID: 236e417e21635cf709fef7826f9b4722@217.113.64.11 CSeq: 103 BYE User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1126-CPO01842 Content-Length: 0 <-------------> [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK29fccab5 (64) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941460-co1126 (58) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "23276647859" ;tag=as060ba4dd (66) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 236e417e21635cf709fef7826f9b4722@217.113.64.11 (55) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 103 BYE (13) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1126-CPO01842 (61) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #219 [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '236e417e21635cf709fef7826f9b4722@217.113.64.11' of Request 103: M atch Not Found Really destroying SIP dialog '236e417e21635cf709fef7826f9b4722@217.113.64.11' Method: INVITE [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0264249918279782@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298763-1168941506@172.16.100.100 From: ;tag=8662 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0264249918279782@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298763-1168941506@172.16.100.100 (54) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=8662 (62) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3 520709 (85) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298763-1168941506@172.16.100.100 localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Content-Length: 0 <------------> [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #224 localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:52] DEBUG[18357]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:58:52] DEBUG[18357]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-0a0ac7b0' [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-0a0ac7b0, SIP callid 45e6c4827c0d0bc80a9897fc074585cd@217.113.6 4.11) [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:3278 sip_hangup: update_call_counter(249918279782) - decrement call limit counter on hangup [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 196.29.163.6:5060: CANCEL sip:249918279782@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport From: "2356221453" ;tag=as7b0b0b1b To: Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #226 Scheduling destruction of SIP dialog '45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:58:52] DEBUG[18357]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-0a0ac7b0 [Jan 16 10:58:52] DEBUG[18357]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:58:52] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 16 10:58:52] DEBUG[18357]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0264249918279782,1) exited non-zero on 'SIP/5060-09ec46e0' == Spawn extension (default, 0264249918279782, 1) exited non-zero on 'SIP/5060-09ec46e0' [Jan 16 10:58:52] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 16 10:58:52] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) [Jan 16 10:58:52] DEBUG[18357]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:58:52] DEBUG[18357]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09ec46e0' [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09ec46e0, SIP callid 6994090221311298763-1168941506@172.16.100.100) [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:58:52] DEBUG[18357]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:58:52] DEBUG[18357]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09ec46e0 [Jan 16 10:58:52] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:52] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:52] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist From: "2356221453" ;tag=as7b0b0b1b To: ;tag=648369a7 CSeq: 102 CANCEL Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport=5060 Warning: 399 sx3000 "Call Leg Id doesn't match" Content-Length: 0 <-------------> [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 481 Call/Transaction Does Not Exist (43) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356221453" ;tag=as7b0b0b1b (64) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=648369a7 (48) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 CANCEL (16) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11 (55) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7620d752;rport=5060 (69) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Warning: 399 sx3000 "Call Leg Id doesn't match" (47) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #226 [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '45e6c4827c0d0bc80a9897fc074585cd@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:52] WARNING[18326]: chan_sip.c:12212 handle_response: Remote host can't match request CANCEL to call '45e6c4827c0d0bc80a9897fc074585cd@217 .113.64.11'. Giving up. [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1171468a To: ;tag=t1168941488-co1272 From: "2425323347" ;tag=as6521d55e Call-ID: 4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1272-CPO02365 Content-Length: 0 <-------------> [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1171468a (64) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941488-co1272 (58) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as6521d55e (64) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11 (55) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1272-CPO02365 (61) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #222 [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11' of Request 102: M atch Not Found localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1171468a To: ;tag=t1168941488-co1272 From: "2425323347" ;tag=as6521d55e Call-ID: 4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1272-CPO02365 Content-Length: 0 <-------------> [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1171468a (64) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941488-co1272 (58) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as6521d55e (64) [Jan 16 10:58:52] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11 (55) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1272-CPO02365 (61) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11' of Request 102: M atch Found [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2204489001@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1171468a;rport From: "2425323347" ;tag=as6521d55e To: ;tag=t1168941488-co1272 Contact: Call-ID: 4e1f2e773ca0de775d54ef0957ee6eed@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:52] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:52] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:52] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205547006@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298823-1168941533@172.16.100.100 From: ;tag=4544 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d07-ac106464-3520748 Contact: sip:972354168@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3369 3369 IN IP4 10.57.3.26 s=- c=IN IP4 10.57.3.26 t=0 0 m=audio 42380 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205547006@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298823-1168941533@172.16.100.100 (54) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4544 (61) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d07-ac106464-3 520748 (85) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:972354168@172.16.100.100:5060;user=phone (53) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3369 3369 IN IP4 10.57.3.26 (40) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.26 (19) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42380 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298823-1168941533@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298823-1168941533@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.26:42380 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.26:42380 [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205547006 in default (domain 10.100.20.11) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:972354168@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a1383c0: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d07-ac106464-3520748;received=172.16.100.100 From: ;tag=4544 To: Call-ID: 6994090221311298823-1168941533@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:53] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a1383c0 [Jan 16 10:58:53] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:53] DEBUG[18366]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432205547006@default:1] Dial("SIP/5060-0a1383c0", "SIP/2205547006@216.226.69.244||t") in new stack [Jan 16 10:58:53] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:53] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:53] DEBUG[18366]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432205547006-1. [Jan 16 10:58:53] DEBUG[18366]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:53] DEBUG[18366]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:53] DEBUG[18366]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:2804 sip_call: Outgoing Call for 2205547006 [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 18754 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205547006@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2776ee23;rport (64) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 2: From: "972354168" ;tag=as196e7d36 (62) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 198614684ac2b2d221e650fc315b4433@217.113.64.11 (55) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:53 GMT (35) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: m=audio 18754 RTP/AVP 18 (24) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205547006@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2776ee23;rport From: "972354168" ;tag=as196e7d36 To: Contact: Call-ID: 198614684ac2b2d221e650fc315b4433@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 18754 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #228 -- Called 2205547006@216.226.69.244 [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #224 (1) SIP/2.0 - 1 [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #224)) Retransmitting #1 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2776ee23 To: From: "972354168" ;tag=as196e7d36 Call-ID: 198614684ac2b2d221e650fc315b4433@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1516-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2776ee23 (64) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972354168" ;tag=as196e7d36 (62) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 198614684ac2b2d221e650fc315b4433@217.113.64.11 (55) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1516-CPO00000 (58) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #228 - INVITE (got response) [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '198614684ac2b2d221e650f c315b4433@217.113.64.11' Request 102: Found [Jan 16 10:58:53] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:53] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14830 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:53] DEBUG[18366]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d07-ac106464-3520748;received=172.16.100.100 From: ;tag=4544 To: ;tag=as10411cd3 Call-ID: 6994090221311298823-1168941533@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14830 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:53] DEBUG[18366]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:53] DEBUG[18366]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:53] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:53] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2776ee23 To: ;tag=t1168941533-co1516 From: "972354168" ;tag=as196e7d36 Call-ID: 198614684ac2b2d221e650fc315b4433@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1516-CPO00292 Content-Type: application/sdp Content-Length: 228 v=0 o=- 245615449 245615449 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 41372 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2776ee23 (64) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941533-co1516 (58) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972354168" ;tag=as196e7d36 (62) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 198614684ac2b2d221e650fc315b4433@217.113.64.11 (55) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1516-CPO00292 (61) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 245615449 245615449 IN IP4 216.226.69.244 (45) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 41372 RTP/AVP 18 101 (28) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '198614684ac2b2d221e650f c315b4433@217.113.64.11' Request 102: Found [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:41372 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f35b90 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:41372 [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: -- SIP/216.226.69.244-09f35b90 is making progress passing it to SIP/5060-0a1383c0 SIP TIMER: Rescheduling retransmission #224 (2) SIP/2.0 - 1 [Jan 16 10:58:54] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #224)) Retransmitting #2 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:58:54] DEBUG[18366]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:54] DEBUG[18366]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:54] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:54] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:55] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #224 (3) SIP/2.0 - 1 [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #224)) Retransmitting #3 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '3bbb90e34d8d70c2227280ac11af4f94@217.113.64.11' [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 3bbb90e34d8d70c2227280ac11af4f94@217.113.64.11 Really destroying SIP dialog '3bbb90e34d8d70c2227280ac11af4f94@217.113.64.11' Method: INVITE [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 180 Ringing From: "2356408270" ;tag=as1376ce66 To: ;tag=43f067f1 CSeq: 102 INVITE Call-ID: 217bc3c0139062936630151f77da5ed9@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK131f9f97;rport=5060 Contact: Content-Length: 151 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 10595228 10595228 IN IP4 10.10.13.4 s=Sip Call =IN IP4 196.29.163.6 t=0 0 m=audio 37088 RTP/AVP 18 a=rtpmap:18 G729/8000 <-------------> [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356408270" ;tag=as1376ce66 (64) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=43f067f1 (48) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 217bc3c0139062936630151f77da5ed9@217.113.64.11 (55) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK131f9f97;rport=5060 (69) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (56) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 151 (19) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=HuaweiSoftX3000 10595228 10595228 IN IP4 10.10.13.4 (53) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=Sip Call (10) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 196.29.163.6 (21) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 37088 RTP/AVP 18 (24) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) --- (9 headers 7 lines) --- [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '217bc3c0139062936630151 f77da5ed9@217.113.64.11' Request 102: Found [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 180 to standard invite [Jan 16 10:58:56] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-0a134458 Found RTP audio format 18 [Jan 16 10:58:56] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 16 10:58:56] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 196.29.163.6:37088 [Jan 16 10:58:56] DEBUG[18321]: channel.c:943 channel_find_locked: Avoiding initial deadlock for channel '0xa046448' Found description format G729 for ID 18 [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/196.29.163.6-0a134458 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 196.29.163.6:37088 [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:56] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/196.29.163.6-0a134458 is ringing localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cfd-ac106464-3520739;received=172.16.100.100 From: ;tag=19565 To: ;tag=as3c8f81fb Call-ID: 6994090221311298813-1168941530@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- SIP/196.29.163.6-0a134458 is making progress passing it to SIP/5060-0a1304f0 [Jan 16 10:58:56] DEBUG[18365]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:56] DEBUG[18365]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:56] DEBUG[18365]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 13906 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:56] DEBUG[18365]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:56] DEBUG[18365]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cfd-ac106464-3520739;received=172.16.100.100 From: ;tag=19565 To: ;tag=as3c8f81fb Call-ID: 6994090221311298813-1168941530@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 13906 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:56] DEBUG[18365]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:56] DEBUG[18365]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 6 (Ringing) [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:56] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:57] DEBUG[18365]: rtp.c:870 ast_rtcp_read: Got RTCP report of 28 bytes [Jan 16 10:58:57] DEBUG[18365]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:57] DEBUG[18365]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:57] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK23e28bc4 To: ;tag=t1168941520-co1455 From: "004769838857" ;tag=as1f5067ce Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1271377745 1271377745 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 52212 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK23e28bc4 (64) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941520-co1455 (58) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "004769838857" ;tag=as1f5067ce (68) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 (55) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 (61) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1271377745 1271377745 IN IP4 216.226.69.244 (47) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 52212 RTP/AVP 18 101 (28) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '126a554e5a5c424454784837291c97de@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:52212 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09ddd560 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.19.106.196:52212 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 126a554e5a5c424454784837291c97de@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2209825337@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK55bc9f33;rport From: "004769838857" ;tag=as1f5067ce To: ;tag=t1168941520-co1455 Contact: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:57] DEBUG[18360]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09ddd560 -- SIP/216.226.69.244-09ddd560 answered SIP/5060-09d74158 [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:57] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:57] DEBUG[18360]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d74158 [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09d74158 [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:57] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11656 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520725;received=172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 Call-ID: 6994090221311298781-1168941520@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11656 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:57] DEBUG[18360]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #234 [Jan 16 10:58:57] DEBUG[18360]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298781-1168941520@172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520725 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298781-1168941520@172.16.100.100 (54) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=15022 (65) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as42a70485 (67) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3 520725 (85) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #234 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298781-1168941520@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3b9ac359 To: ;tag=t1168941528-co1492 From: "933122088" ;tag=as419958cb Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1492-CPO00194 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1205031023 1205031023 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 53156 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3b9ac359 (64) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941528-co1492 (58) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "933122088" ;tag=as419958cb (62) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 (55) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1492-CPO00194 (61) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1205031023 1205031023 IN IP4 216.226.69.244 (47) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 53156 RTP/AVP 18 101 (28) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '79458a6c010104ae25e6ffbf3851550d@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:53156 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f50520 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.19.106.196:53156 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2209935770@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK43176963;rport From: "933122088" ;tag=as419958cb To: ;tag=t1168941528-co1492 Contact: Call-ID: 79458a6c010104ae25e6ffbf3851550d@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:57] DEBUG[18362]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09f50520 -- SIP/216.226.69.244-09f50520 answered SIP/5060-09dc79f0 [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:57] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:57] DEBUG[18362]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09dc79f0 [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09dc79f0 [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:57] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:57] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14286 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3520731;received=172.16.100.100 From: ;tag=1818 To: ;tag=as225a1882 Call-ID: 6994090221311298795-1168941527@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14286 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------>> [Jan 16 10:58:57] DEBUG[18362]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #235 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432209935770@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298795-1168941527@172.16.100.100 From: ;tag=1818 To: ;tag=as225a1882 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3520731 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432209935770@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298795-1168941527@172.16.100.100 (54) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1818 (61) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as225a1882 (67) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ceb-ac106464-3 520731 (85) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #235 [Jan 16 10:58:57] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298795-1168941527@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18362]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:57] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK04843c9a To: ;tag=t1168941529-co1503 From: "972402839" ;tag=as2f3d0882 Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1503-CPO00225 Content-Type: application/sdp Content-Length: 226 v=0 o=- 48898701 48898701 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 42620 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK04843c9a (64) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941529-co1503 (58) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972402839" ;tag=as2f3d0882 (62) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 (55) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1503-CPO00225 (61) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 226 (19) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 48898701 48898701 IN IP4 216.226.69.244 (43) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42620 RTP/AVP 18 101 (28) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:42620 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09f51aa0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:42620 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2204484010@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK490bfbca;rport From: "972402839" ;tag=as2f3d0882 To: ;tag=t1168941529-co1503 Contact: Call-ID: 3b86aa9e1a51c7a672149ed3529fb28c@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:58:58] DEBUG[18363]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09f51aa0 -- SIP/216.226.69.244-09f51aa0 answered SIP/5060-09d94ff0 [Jan 16 10:58:58] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:58:58] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:58:58] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:58:58] DEBUG[18363]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d94ff0 [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09d94ff0 [Jan 16 10:58:58] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:58] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:58] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 14642 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3520733;received=172.16.100.100 From: ;tag=2727 To: ;tag=as3756653a Call-ID: 6994090221311298799-1168941529@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 14642 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:58] DEBUG[18363]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #236 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432204484010@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298799-1168941529@172.16.100.100 From: ;tag=2727 To: ;tag=as3756653a CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3520733 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432204484010@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298799-1168941529@172.16.100.100 (54) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=2727 (61) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as3756653a (67) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cef-ac106464-3 520733 (85) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #236 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298799-1168941529@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:58:58] DEBUG[18363]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:58] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:58] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:58] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:58] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205667389@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298835-1168941538@172.16.100.100 From: ;tag=23199 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3520755 Contact: sip:934690046@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3850 3850 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 45872 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205667389@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298835-1168941538@172.16.100.100 (54) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=23199 (62) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3 520755 (85) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:934690046@172.16.100.100:5060;user=phone (53) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3850 3850 IN IP4 10.57.3.22 (40) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 45872 RTP/AVP 18 0 101 13 (33) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298835-1168941538@172.16.100.100 - INVITE (With RT P) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298835-1168941538@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:45872 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.22:45872 [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205667389 in default (domain 10.100.20.11) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:934690046@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:58:58] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09fb24f8: New call is still down.... Trying... localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3520755;received=172.16.100.100 From: ;tag=23199 To: Call-ID: 6994090221311298835-1168941538@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:58:58] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09fb24f8 [Jan 16 10:58:58] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:58:58] DEBUG[18367]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432205667389@default:1] Dial("SIP/5060-09fb24f8", "SIP/2205667389@216.226.69.244||t") in new stack [Jan 16 10:58:58] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:58:58] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:58:58] DEBUG[18367]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432205667389-1. [Jan 16 10:58:58] DEBUG[18367]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:58:58] DEBUG[18367]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:58:58] DEBUG[18367]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:2804 sip_call: Outgoing Call for 2205667389 [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 10448 Adding codec 0x100 (g729) to SDP [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205667389@216.226.69.244 SIP/2.0 (44) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1beb9c47;rport (64) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 2: From: "934690046" ;tag=as46cc04f6 (62) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 (55) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:58:58 GMT (35) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: m=audio 10448 RTP/AVP 18 (24) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205667389@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1beb9c47;rport From: "934690046" ;tag=as46cc04f6 To: Contact: Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:58:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 10448 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:58:58] DEBUG[18367]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #237 -- Called 2205667389@216.226.69.244 [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:58] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1beb9c47 To: From: "934690046" ;tag=as46cc04f6 Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1541-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1beb9c47 (64) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "934690046" ;tag=as46cc04f6 (62) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 (55) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1541-CPO00000 (58) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #237 - INVITE (got response) [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '15cb97af09d6b84f2345358 566576ca3@217.113.64.11' Request 102: Found [Jan 16 10:58:59] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18367]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:58:59] DEBUG[18367]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:58:59] DEBUG[18367]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:58:59] DEBUG[18367]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 10404 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:58:59] DEBUG[18367]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:58:59] DEBUG[18367]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3520755;received=172.16.100.100 From: ;tag=23199 To: ;tag=as31e8d0e9 Call-ID: 6994090221311298835-1168941538@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 10404 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:58:59] DEBUG[18367]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:58:59] DEBUG[18367]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:59] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:58:59] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:58:59] DEBUG[18367]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1beb9c47 To: ;tag=t1168941539-co1541 From: "934690046" ;tag=as46cc04f6 Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1541-CPO00416 Content-Type: application/sdp Content-Length: 230 v=0 o=- 2075814817 2075814817 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43292 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1beb9c47 (64) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941539-co1541 (58) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "934690046" ;tag=as46cc04f6 (62) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 (55) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1541-CPO00416 (61) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 2075814817 2075814817 IN IP4 216.226.69.244 (47) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43292 RTP/AVP 18 101 (28) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '15cb97af09d6b84f2345358 566576ca3@217.113.64.11' Request 102: Found [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43292 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09db30f0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43292 [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-09db30f0 is making progress passing it to SIP/5060-09fb24f8 [Jan 16 10:59:00] DEBUG[18367]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:00] DEBUG[18367]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209913229@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298837-1168941539@172.16.100.100 From: ;tag=7270 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757 Contact: sip:2425323347@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 226 v=0 o=MG4000|2.0 51535 51535 IN IP4 10.57.2.137 s=- c=IN IP4 10.57.2.137 t=0 0 m=audio 62140 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209913229@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298837-1168941539@172.16.100.100 (54) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=7270 (62) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3 520757 (85) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2425323347@172.16.100.100:5060;user=phone (54) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 226 (19) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 51535 51535 IN IP4 10.57.2.137 (43) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.137 (20) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 62140 RTP/AVP 18 0 101 13 (33) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298837-1168941539@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298837-1168941539@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.137:62140 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.137:62140 [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209913229 in default (domain 10.100.20.11) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2425323347@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a1605a0: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757;received=172.16.100.100 From: ;tag=7270 To: Call-ID: 6994090221311298837-1168941539@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:00] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a1605a0 [Jan 16 10:59:00] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:00] DEBUG[18368]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432209913229@default:1] Dial("SIP/5060-0a1605a0", "SIP/2209913229@216.226.69.244||t") in new stack [Jan 16 10:59:00] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:00] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:00] DEBUG[18368]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432209913229-1. [Jan 16 10:59:00] DEBUG[18368]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:00] DEBUG[18368]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:00] DEBUG[18368]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:2804 sip_call: Outgoing Call for 2209913229 [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 15596 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209913229@216.226.69.244 SIP/2.0 (44) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1ee4680a;rport (64) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 2: From: "2425323347" ;tag=as4f2ed6be (64) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 (55) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:00 GMT (35) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: m=audio 15596 RTP/AVP 18 (24) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209913229@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1ee4680a;rport From: "2425323347" ;tag=as4f2ed6be To: Contact: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 15596 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #241 -- Called 2209913229@216.226.69.244 [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #224 (4) SIP/2.0 - 1 [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #224)) Retransmitting #4 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '6994090221311298757-1168941504@172.16.100.100' [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 6994090221311298757-1168941504@172.16.100.100 Really destroying SIP dialog '6994090221311298757-1168941504@172.16.100.100' Method: ACK [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a To: From: "2425323347" ;tag=as4f2ed6be Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1542-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a (64) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as4f2ed6be (64) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 (55) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1542-CPO00000 (58) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #241 - INVITE (got response) [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '31bb61106db989506647338 a4aa16842@217.113.64.11' Request 102: Found [Jan 16 10:59:00] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18368]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19044 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:00] DEBUG[18368]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757;received=172.16.100.100 From: ;tag=7270 To: ;tag=as23224831 Call-ID: 6994090221311298837-1168941539@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:00] DEBUG[18368]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:00] DEBUG[18368]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:00] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a To: ;tag=t1168941540-co1542 From: "2425323347" ;tag=as4f2ed6be Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1542-CPO00427 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1218714249 1218714249 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 53988 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a (64) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941540-co1542 (58) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as4f2ed6be (64) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 (55) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1542-CPO00427 (61) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1218714249 1218714249 IN IP4 216.226.69.244 (47) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 53988 RTP/AVP 18 101 (28) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '31bb61106db989506647338 a4aa16842@217.113.64.11' Request 102: Found [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:53988 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0a171b10 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.19.106.196:53988 [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-0a171b10 is making progress passing it to SIP/5060-0a1605a0 [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18365]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18368]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:01] DEBUG[18368]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18365]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' [Jan 16 10:59:01] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11 Really destroying SIP dialog '70bc55591d2ff40e0aaf62d77c34e44e@217.113.64.11' Method: INVITE [Jan 16 10:59:01] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18368]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:01] DEBUG[18368]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:02] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:02] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:03] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:03] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:03] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:03] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18365]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #224 (5) SIP/2.0 - 1 [Jan 16 10:59:04] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #224)) Retransmitting #5 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18367]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:04] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:04] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18368]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:05] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:05] DEBUG[18368]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432209913229@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298837-1168941539@172.16.100.100 From: ;tag=7270 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432209913229@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298837-1168941539@172.16.100.100 (54) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=7270 (62) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3 520757 (85) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298837-1168941539@172.16.100.100 localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757;received=172.16.100.100 From: ;tag=7270 To: ;tag=as23224831 Call-ID: 6994090221311298837-1168941539@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #245 localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757;received=172.16.100.100 From: ;tag=7270 To: ;tag=as23224831 Call-ID: 6994090221311298837-1168941539@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:05] DEBUG[18368]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:59:05] DEBUG[18368]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-0a171b10' [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-0a171b10, SIP callid 31bb61106db989506647338a4aa16842@217.113 .64.11) [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:3278 sip_hangup: update_call_counter(2209913229) - decrement call limit counter on hangup [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '31bb61106db989506647338a4aa16842@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '31bb61106db989506647338a4aa16842@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2209913229@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1ee4680a;rport From: "2425323347" ;tag=as4f2ed6be To: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #247 Scheduling destruction of SIP dialog '31bb61106db989506647338a4aa16842@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:05] DEBUG[18368]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0a171b10 [Jan 16 10:59:05] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:59:05] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:59:05] DEBUG[18368]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:59:05] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:59:05] DEBUG[18368]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432209913229,1) exited non-zero on 'SIP/5060-0a1605a0' == Spawn extension (default, 0432209913229, 1) exited non-zero on 'SIP/5060-0a1605a0' [Jan 16 10:59:05] DEBUG[18368]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:59:05] DEBUG[18368]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-0a1605a0' [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-0a1605a0, SIP callid 6994090221311298837-1168941539@172.16.100.100) [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:59:05] DEBUG[18368]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:59:05] DEBUG[18368]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a1605a0 [Jan 16 10:59:05] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:05] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:05] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432209913229@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298837-1168941539@172.16.100.100 From: ;tag=7270 To: ;tag=as23224831 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3520757 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432209913229@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298837-1168941539@172.16.100.100 (54) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=7270 (62) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as23224831 (67) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d15-ac106464-3 520757 (85) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #245 [Jan 16 10:59:05] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298837-1168941539@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311298837-1168941539@172.16.100.100' Method: ACK [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:05] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a To: ;tag=t1168941540-co1542 From: "2425323347" ;tag=as4f2ed6be Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1542-CPO00427 Content-Length: 0 <-------------> [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a (64) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941540-co1542 (58) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as4f2ed6be (64) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 (55) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1542-CPO00427 (61) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #247 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '31bb61106db989506647338a4aa16842@217.113.64.11' of Request 102: M atch Not Found localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a To: ;tag=t1168941540-co1542 From: "2425323347" ;tag=as4f2ed6be Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1542-CPO00427 Content-Length: 0 <-------------> [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1ee4680a (64) [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941540-co1542 (58) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as4f2ed6be (64) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 (55) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1542-CPO00427 (61) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '31bb61106db989506647338a4aa16842@217.113.64.11' of Request 102: M atch Found [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2209913229@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1ee4680a;rport From: "2425323347" ;tag=as4f2ed6be To: ;tag=t1168941540-co1542 Contact: Call-ID: 31bb61106db989506647338a4aa16842@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18367]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432207799289@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298843-1168941546@172.16.100.100 From: ;tag=26834 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1b-ac106464-3520761 Contact: sip:4532322555@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 4069 4069 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 39724 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432207799289@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298843-1168941546@172.16.100.100 (54) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=26834 (63) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1b-ac106464-3 520761 (85) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:4532322555@172.16.100.100:5060;user=phone (54) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 4069 4069 IN IP4 10.57.3.22 (40) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 39724 RTP/AVP 18 0 101 13 (33) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298843-1168941546@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:06] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298843-1168941546@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:39724 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.22:39724 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432207799289 in default (domain 10.100.20.11) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:4532322555@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09f01758: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1b-ac106464-3520761;received=172.16.100.100 From: ;tag=26834 To: Call-ID: 6994090221311298843-1168941546@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:06] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09f01758 [Jan 16 10:59:06] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:06] DEBUG[18369]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432207799289@default:1] Dial("SIP/5060-09f01758", "SIP/2207799289@216.226.69.244||t") in new stack [Jan 16 10:59:06] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:06] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:06] DEBUG[18369]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432207799289-1. [Jan 16 10:59:06] DEBUG[18369]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:06] DEBUG[18369]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:06] DEBUG[18369]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:2804 sip_call: Outgoing Call for 2207799289 [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 15316 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2207799289@216.226.69.244 SIP/2.0 (44) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5b75ada5;rport (64) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 2: From: "4532322555" ;tag=as6d2feea1 (64) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 71c97aa5150eb02b7b661cce06bba94e@217.113.64.11 (55) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:06 GMT (35) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: m=audio 15316 RTP/AVP 18 (24) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2207799289@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5b75ada5;rport From: "4532322555" ;tag=as6d2feea1 To: Contact: Call-ID: 71c97aa5150eb02b7b661cce06bba94e@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 15316 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:06] DEBUG[18369]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #249 -- Called 2207799289@216.226.69.244 [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432208100493@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298845-1168941546@172.16.100.100 From: ;tag=26834 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1d-ac106464-3520762 Contact: sip:953375013@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 1002 1002 IN IP4 10.57.3.25 s=- c=IN IP4 10.57.3.25 t=0 0 m=audio 31164 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432208100493@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298845-1168941546@172.16.100.100 (54) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=26834 (62) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1d-ac106464-3 520762 (85) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:953375013@172.16.100.100:5060;user=phone (53) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 1002 1002 IN IP4 10.57.3.25 (40) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.25 (19) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 31164 RTP/AVP 18 0 101 13 (33) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298845-1168941546@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298845-1168941546@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.25:31164 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.25:31164 [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432208100493 in default (domain 10.100.20.11) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:953375013@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:06] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a170590: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1d-ac106464-3520762;received=172.16.100.100 From: ;tag=26834 To: Call-ID: 6994090221311298845-1168941546@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:06] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a170590 [Jan 16 10:59:06] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:06] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:06] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:06] DEBUG[18370]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432208100493@default:1] Dial("SIP/5060-0a170590", "SIP/2208100493@216.226.69.244||t") in new stack [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:06] DEBUG[18370]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432208100493-1. [Jan 16 10:59:06] DEBUG[18370]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:06] DEBUG[18370]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:06] DEBUG[18370]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:2804 sip_call: Outgoing Call for 2208100493 [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 15478 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2208100493@216.226.69.244 SIP/2.0 (44) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2292d63c;rport (64) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 2: From: "953375013" ;tag=as0f2de13d (62) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 0a44817b02d053621fee990d51a9fa1e@217.113.64.11 (55) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:06 GMT (35) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: m=audio 15478 RTP/AVP 18 (24) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2208100493@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2292d63c;rport From: "953375013" ;tag=as0f2de13d To: Contact: Call-ID: 0a44817b02d053621fee990d51a9fa1e@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 15478 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:06] DEBUG[18370]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #251 -- Called 2208100493@216.226.69.244 [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:06] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5b75ada5 To: From: "4532322555" ;tag=as6d2feea1 Call-ID: 71c97aa5150eb02b7b661cce06bba94e@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1563-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5b75ada5 (64) [Jan 16 10:59:07] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "4532322555" ;tag=as6d2feea1 (64) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 71c97aa5150eb02b7b661cce06bba94e@217.113.64.11 (55) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1563-CPO00000 (58) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #249 - INVITE (got response) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '71c97aa5150eb02b7b661cc e06bba94e@217.113.64.11' Request 102: Found [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:07] DEBUG[18338]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18369]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:59:07] DEBUG[18369]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:07] DEBUG[18369]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:07] DEBUG[18369]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15328 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:07] DEBUG[18369]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:07] DEBUG[18369]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1b-ac106464-3520761;received=172.16.100.100 From: ;tag=26834 To: ;tag=as651a73be Call-ID: 6994090221311298843-1168941546@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15328 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:07] DEBUG[18369]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:07] DEBUG[18369]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2292d63c To: From: "953375013" ;tag=as0f2de13d Call-ID: 0a44817b02d053621fee990d51a9fa1e@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1564-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2292d63c (64) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "953375013" ;tag=as0f2de13d (62) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0a44817b02d053621fee990d51a9fa1e@217.113.64.11 (55) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1564-CPO00000 (58) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #251 - INVITE (got response) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0a44817b02d053621fee990 d51a9fa1e@217.113.64.11' Request 102: Found [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18370]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:07] DEBUG[18370]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:07] DEBUG[18370]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:07] DEBUG[18370]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 17092 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:07] DEBUG[18370]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:07] DEBUG[18370]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1d-ac106464-3520762;received=172.16.100.100 From: ;tag=26834 To: ;tag=as64b896dc Call-ID: 6994090221311298845-1168941546@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 17092 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:07] DEBUG[18370]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:07] DEBUG[18370]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18369]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298847-1168941547@172.16.100.100 From: ;tag=10904 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1f-ac106464-3520764 Contact: sip:2356756282@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 224 v=0 o=MG4000|2.0 10040 10040 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 63320 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249734822305@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298847-1168941547@172.16.100.100 (54) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=10904 (63) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1f-ac106464-3 520764 (85) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356756282@172.16.100.100:5060;user=phone (54) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 224 (19) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 10040 10040 IN IP4 10.57.2.121 (43) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 63320 RTP/AVP 18 101 13 (31) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298847-1168941547@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298847-1168941547@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:63320 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.121:63320 [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249734822305 in default (domain 10.100.20.11) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356756282@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:07] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09ec0480: New call is still down.... Trying... localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d1f-ac106464-3520764;received=172.16.100.100 From: ;tag=10904 To: Call-ID: 6994090221311298847-1168941547@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:07] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09ec0480 [Jan 16 10:59:07] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:07] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:07] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:07] DEBUG[18371]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0264249734822305@default:1] Dial("SIP/5060-09ec0480", "SIP/249734822305@196.29.163.6||t") in new stack [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:07] DEBUG[18371]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0264249734822305-1. [Jan 16 10:59:07] DEBUG[18371]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:07] DEBUG[18371]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:07] DEBUG[18371]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:2804 sip_call: Outgoing Call for 249734822305 [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 16866 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249734822305@196.29.163.6 SIP/2.0 (44) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3209708c;rport (64) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 2: From: "2356756282" ;tag=as6ddb37ad (64) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 001f622a10c230d73b1c9ae457dd3ce6@217.113.64.11 (55) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:07 GMT (35) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: m=audio 16866 RTP/AVP 18 (24) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3209708c;rport From: "2356756282" ;tag=as6ddb37ad To: Contact: Call-ID: 001f622a10c230d73b1c9ae457dd3ce6@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16866 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:07] DEBUG[18371]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #255 -- Called 249734822305@196.29.163.6 [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:07] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '6994090221311298743-1168941499@172.16.100.100' [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 6994090221311298743-1168941499@172.16.100.100 Really destroying SIP dialog '6994090221311298743-1168941499@172.16.100.100' Method: ACK [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18370]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2292d63c To: ;tag=t1168941547-co1564 From: "953375013" ;tag=as0f2de13d Call-ID: 0a44817b02d053621fee990d51a9fa1e@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1564-CPO00524 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1870436184 1870436184 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40192 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK2292d63c (64) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941547-co1564 (58) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "953375013" ;tag=as0f2de13d (62) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0a44817b02d053621fee990d51a9fa1e@217.113.64.11 (55) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1564-CPO00524 (61) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 1870436184 1870436184 IN IP4 216.226.69.244 (47) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 40192 RTP/AVP 18 101 (28) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0a44817b02d053621fee990 d51a9fa1e@217.113.64.11' Request 102: Found [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40192 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0a03e5b0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:40192 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-0a03e5b0 is making progress passing it to SIP/5060-0a170590 [Jan 16 10:59:08] DEBUG[18370]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:08] DEBUG[18370]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432208101110@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298855-1168941548@172.16.100.100 From: ;tag=27743 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765 Contact: sip:972370597@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 222 v=0 o=MG4000|2.0 3517 3517 IN IP4 10.57.3.26 s=- c=IN IP4 10.57.3.26 t=0 0 m=audio 49064 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432208101110@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298855-1168941548@172.16.100.100 (54) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=27743 (62) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3 520765 (85) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:972370597@172.16.100.100:5060;user=phone (53) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3517 3517 IN IP4 10.57.3.26 (40) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.26 (19) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 49064 RTP/AVP 18 0 101 13 (33) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298855-1168941548@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298855-1168941548@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.26:49064 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.3.26:49064 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432208101110 in default (domain 10.100.20.11) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:972370597@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09e99548: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765;received=172.16.100.100 From: ;tag=27743 To: Call-ID: 6994090221311298855-1168941548@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:08] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09e99548 [Jan 16 10:59:08] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:08] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:08] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:08] DEBUG[18372]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432208101110@default:1] Dial("SIP/5060-09e99548", "SIP/2208101110@216.226.69.244||t") in new stack [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:08] DEBUG[18372]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432208101110-1. [Jan 16 10:59:08] DEBUG[18372]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:08] DEBUG[18372]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:08] DEBUG[18372]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:2804 sip_call: Outgoing Call for 2208101110 [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 18510 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2208101110@216.226.69.244 SIP/2.0 (44) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK37cc3953;rport (64) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 2: From: "972370597" ;tag=as12354d03 (62) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 (55) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:08 GMT (35) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: m=audio 18510 RTP/AVP 18 (24) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2208101110@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK37cc3953;rport From: "972370597" ;tag=as12354d03 To: Contact: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 18510 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #258 -- Called 2208101110@216.226.69.244 [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #224 (6) SIP/2.0 - 1 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #224)) Retransmitting #6 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ccb-ac106464-3520709;received=172.16.100.100 From: ;tag=8662 To: ;tag=as0740ad43 Call-ID: 6994090221311298763-1168941506@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18355]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:08] DEBUG[18362]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249915386814@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298853-1168941547@172.16.100.100 From: ;tag=10904 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d25-ac106464-3520766 Contact: sip:2356253529@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 224 v=0 o=MG4000|2.0 10001 10001 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 43636 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249915386814@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298853-1168941547@172.16.100.100 (54) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=10904 (63) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d25-ac106464-3 520766 (85) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356253529@172.16.100.100:5060;user=phone (54) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 224 (19) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 10001 10001 IN IP4 10.57.2.121 (43) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43636 RTP/AVP 18 101 13 (31) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298853-1168941547@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298853-1168941547@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:43636 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.121:43636 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249915386814 in default (domain 10.100.20.11) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356253529@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a1d3438: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d25-ac106464-3520766;received=172.16.100.100 From: ;tag=10904 To: Call-ID: 6994090221311298853-1168941547@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:08] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a1d3438 [Jan 16 10:59:08] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:08] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:08] DEBUG[18373]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0264249915386814@default:1] Dial("SIP/5060-0a1d3438", "SIP/249915386814@196.29.163.6||t") in new stack [Jan 16 10:59:08] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:08] DEBUG[18373]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0264249915386814-1. [Jan 16 10:59:08] DEBUG[18373]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:08] DEBUG[18373]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:08] DEBUG[18373]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:2804 sip_call: Outgoing Call for 249915386814 [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 16014 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249915386814@196.29.163.6 SIP/2.0 (44) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7b1e443e;rport (64) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 2: From: "2356253529" ;tag=as56895a96 (64) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 4c38f31128e42af118b19d1618d0ff40@217.113.64.11 (55) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:08 GMT (35) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: m=audio 16014 RTP/AVP 18 (24) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249915386814@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7b1e443e;rport From: "2356253529" ;tag=as56895a96 To: Contact: Call-ID: 4c38f31128e42af118b19d1618d0ff40@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16014 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:08] DEBUG[18373]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #260 -- Called 249915386814@196.29.163.6 [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356253529" ;tag=as56895a96 To: CSeq: 102 INVITE Call-ID: 4c38f31128e42af118b19d1618d0ff40@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7b1e443e;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356253529" ;tag=as56895a96 (64) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4c38f31128e42af118b19d1618d0ff40@217.113.64.11 (55) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7b1e443e;rport=5060 (69) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #260 - INVITE (got response) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4c38f31128e42af118b19d1 618d0ff40@217.113.64.11' Request 102: Found [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 To: From: "972370597" ;tag=as12354d03 Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1571-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 (64) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972370597" ;tag=as12354d03 (62) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 (55) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1571-CPO00000 (58) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #258 - INVITE (got response) [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6b34a7636ed1c7bb3a938e2 944d80fa4@217.113.64.11' Request 102: Found [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18345]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #255 (1) INVITE - 5 [Jan 16 10:59:08] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #255)) Retransmitting #1 (no NAT) to 196.29.163.6:5060: INVITE sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3209708c;rport From: "2356756282" ;tag=as6ddb37ad To: Contact: Call-ID: 001f622a10c230d73b1c9ae457dd3ce6@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16866 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18366]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:08] DEBUG[18372]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19044 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:08] DEBUG[18372]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765;received=172.16.100.100 From: ;tag=27743 To: ;tag=as34b4d308 Call-ID: 6994090221311298855-1168941548@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:08] DEBUG[18372]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:08] DEBUG[18372]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:08] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0264249915583309@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298789-1168941523@172.16.100.100 From: ;tag=1 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0264249915583309@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298789-1168941523@172.16.100.100 (54) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1 (59) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3 520729 (85) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298789-1168941523@172.16.100.100 localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729;received=172.16.100.100 From: ;tag=1 To: ;tag=as1a0e9361 Call-ID: 6994090221311298789-1168941523@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #263 localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729;received=172.16.100.100 From: ;tag=1 To: ;tag=as1a0e9361 Call-ID: 6994090221311298789-1168941523@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:09] DEBUG[18361]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:59:09] DEBUG[18361]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-09d8e6a0' [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-09d8e6a0, SIP callid 57c2d72d608b13b93f0a972e6f4e6d56@217.113.6 4.11) [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:3278 sip_hangup: update_call_counter(249915583309) - decrement call limit counter on hangup [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 196.29.163.6:5060: CANCEL sip:249915583309@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport From: "2356652452" ;tag=as0034b53c To: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #265 Scheduling destruction of SIP dialog '57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:09] DEBUG[18361]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-09d8e6a0 [Jan 16 10:59:09] DEBUG[18361]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:59:09] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 16 10:59:09] DEBUG[18361]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0264249915583309,1) exited non-zero on 'SIP/5060-09d8a738' == Spawn extension (default, 0264249915583309, 1) exited non-zero on 'SIP/5060-09d8a738' [Jan 16 10:59:09] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 16 10:59:09] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) [Jan 16 10:59:09] DEBUG[18361]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:59:09] DEBUG[18361]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09d8a738' [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09d8a738, SIP callid 6994090221311298789-1168941523@172.16.100.100) [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:59:09] DEBUG[18361]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:59:09] DEBUG[18361]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d8a738 [Jan 16 10:59:09] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:09] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:09] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18372]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 To: ;tag=t1168941548-co1571 From: "972370597" ;tag=as12354d03 Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1571-CPO00554 Content-Type: application/sdp Content-Length: 228 v=0 o=- 248775465 248775465 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43576 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 (64) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941548-co1571 (58) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972370597" ;tag=as12354d03 (62) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 (55) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1571-CPO00554 (61) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 248775465 248775465 IN IP4 216.226.69.244 (45) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43576 RTP/AVP 18 101 (28) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6b34a7636ed1c7bb3a938e2 944d80fa4@217.113.64.11' Request 102: Found [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43576 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0a1c1ba0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43576 [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-0a1c1ba0 is making progress passing it to SIP/5060-09e99548 [Jan 16 10:59:09] DEBUG[18342]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:09] DEBUG[18372]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:09] DEBUG[18372]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 200 OK From: "2356652452" ;tag=as0034b53c To: CSeq: 102 CANCEL Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356652452" ;tag=as0034b53c (64) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 CANCEL (16) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 (55) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 (69) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #265 [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11' of Request 102: M atch Not Found localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 487 Request Terminated From: "2356652452" ;tag=as0034b53c To: ;tag=9969af1a CSeq: 102 INVITE Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356652452" ;tag=as0034b53c (64) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=9969af1a (48) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 (55) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 (69) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11' of Request 102: M atch Found [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249915583309@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport From: "2356652452" ;tag=as0034b53c To: ;tag=9969af1a Contact: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18367]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18359]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:09] DEBUG[18336]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #255 (2) INVITE - 5 [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #255)) Retransmitting #2 (no NAT) to 196.29.163.6:5060: INVITE sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3209708c;rport From: "2356756282" ;tag=as6ddb37ad To: Contact: Call-ID: 001f622a10c230d73b1c9ae457dd3ce6@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16866 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18363]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:09] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 487 Request Terminated From: "2356652452" ;tag=as0034b53c To: ;tag=9969af1a CSeq: 102 INVITE Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 Content-Length: 0 <-------------> [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: From: "2356652452" ;tag=as0034b53c (64) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=9969af1a (48) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 (55) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport=5060 (69) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11' of Request 102: M atch Found [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249915583309@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0382154c;rport From: "2356652452" ;tag=as0034b53c To: ;tag=9969af1a Contact: Call-ID: 57c2d72d608b13b93f0a972e6f4e6d56@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:09] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #263 (1) SIP/2.0 - 1 [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #263)) Retransmitting #1 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729;received=172.16.100.100 From: ;tag=1 To: ;tag=as1a0e9361 Call-ID: 6994090221311298789-1168941523@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18365]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18354]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18369]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INFO sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298781-1168941520@172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 CSeq: 2 INFO Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520770 Supported: timer,100rel Max-Forwards: 70 Content-Length: 11 Content-Type: application/dtmf-relay Signal= # <-------------> [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INFO sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 (59) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298781-1168941520@172.16.100.100 (54) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=15022 (65) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as42a70485 (67) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 2 INFO (12) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3 520770 (85) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 11 (18) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Type: application/dtmf-relay (36) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: Signal= # (9) --- (10 headers 1 lines) --- [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INFO (13) - Command in SIP INFO Receiving INFO! * DTMF-relay event received: # localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520770;received=172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 Call-ID: 6994090221311298781-1168941520@172.16.100.100 CSeq: 2 INFO User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209980909@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298857-1168941550@172.16.100.100 From: ;tag=28651 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d29-ac106464-3520771 Contact: sip:243851213877@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 224 v=0 o=MG4000|2.0 36560 36560 IN IP4 10.57.2.243 s=- c=IN IP4 10.57.2.243 t=0 0 m=audio 33408 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209980909@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298857-1168941550@172.16.100.100 (54) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=28651 (65) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d29-ac106464-3 520771 (85) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:243851213877@172.16.100.100:5060;user=phone (56) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 224 (19) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 36560 36560 IN IP4 10.57.2.243 (43) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.243 (20) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 33408 RTP/AVP 18 101 13 (31) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298857-1168941550@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298857-1168941550@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.243:33408 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.243:33408 [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209980909 in default (domain 10.100.20.11) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:243851213877@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-0a1fff08: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d29-ac106464-3520771;received=172.16.100.100 From: ;tag=28651 To: Call-ID: 6994090221311298857-1168941550@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:10] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0a1fff08 [Jan 16 10:59:10] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:10] DEBUG[18374]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' -- Executing [0432209980909@default:1] Dial("SIP/5060-0a1fff08", "SIP/2209980909@216.226.69.244||t") in new stack [Jan 16 10:59:10] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:10] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:10] DEBUG[18374]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432209980909-1. [Jan 16 10:59:10] DEBUG[18374]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:10] DEBUG[18374]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:10] DEBUG[18374]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:2804 sip_call: Outgoing Call for 2209980909 [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 15760 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209980909@216.226.69.244 SIP/2.0 (44) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK641931be;rport (64) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 2: From: "243851213877" ;tag=as6a3615af (68) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 1da2ebd934fc944f608dadd04bc25533@217.113.64.11 (55) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:10 GMT (35) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: m=audio 15760 RTP/AVP 18 (24) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209980909@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK641931be;rport From: "243851213877" ;tag=as6a3615af To: Contact: Call-ID: 1da2ebd934fc944f608dadd04bc25533@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 15760 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:10] DEBUG[18374]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #268 -- Called 2209980909@216.226.69.244 [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed <--- SIP read from 172.16.100.100:5060 ---> INFO sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298781-1168941520@172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 CSeq: 3 INFO Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520772 Supported: timer,100rel Max-Forwards: 70 Content-Length: 11 Content-Type: application/dtmf-relay Signal= # <-------------> [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INFO sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 (59) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298781-1168941520@172.16.100.100 (54) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=15022 (65) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as42a70485 (67) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 3 INFO (12) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3 520772 (85) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 11 (18) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Type: application/dtmf-relay (36) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: Signal= # (9) --- (10 headers 1 lines) --- [Jan 16 10:59:10] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INFO (13) - Command in SIP INFO Receiving INFO! * DTMF-relay event received: # <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520772;received=172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 Call-ID: 6994090221311298781-1168941520@172.16.100.100 CSeq: 3 INFO User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:10] DEBUG[18350]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18335]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 16 10:59:11] DEBUG[18372]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432208101110@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298855-1168941548@172.16.100.100 From: ;tag=27743 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432208101110@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298855-1168941548@172.16.100.100 (54) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=27743 (62) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3 520765 (85) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298855-1168941548@172.16.100.100 <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765;received=172.16.100.100 From: ;tag=27743 To: ;tag=as34b4d308 Call-ID: 6994090221311298855-1168941548@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #270 <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765;received=172.16.100.100 From: ;tag=27743 To: ;tag=as34b4d308 Call-ID: 6994090221311298855-1168941548@172.16.100.100 CSeq: 1 CANCEL User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:11] DEBUG[18372]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:59:11] DEBUG[18372]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-0a1c1ba0' [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-0a1c1ba0, SIP callid 6b34a7636ed1c7bb3a938e2944d80fa4@217.113 .64.11) [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:3278 sip_hangup: update_call_counter(2208101110) - decrement call limit counter on hangup [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2208101110@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK37cc3953;rport From: "972370597" ;tag=as12354d03 To: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #272 Scheduling destruction of SIP dialog '6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:11] DEBUG[18372]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0a1c1ba0 [Jan 16 10:59:11] DEBUG[18372]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:59:11] DEBUG[18372]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432208101110,1) exited non-zero on 'SIP/5060-09e99548' == Spawn extension (default, 0432208101110, 1) exited non-zero on 'SIP/5060-09e99548' [Jan 16 10:59:11] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:59:11] DEBUG[18372]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:59:11] DEBUG[18372]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09e99548' [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09e99548, SIP callid 6994090221311298855-1168941548@172.16.100.100) [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:59:11] DEBUG[18372]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 16 10:59:11] DEBUG[18372]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09e99548 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:11] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432208101110@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298855-1168941548@172.16.100.100 From: ;tag=27743 To: ;tag=as34b4d308 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3520765 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432208101110@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298855-1168941548@172.16.100.100 (54) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=27743 (62) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as34b4d308 (67) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d27-ac106464-3 520765 (85) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #270 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298855-1168941548@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311298855-1168941548@172.16.100.100' Method: ACK [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #263 (2) SIP/2.0 - 1 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #263)) Retransmitting #2 (no NAT) to 172.16.100.100:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0ce5-ac106464-3520729;received=172.16.100.100 From: ;tag=1 To: ;tag=as1a0e9361 Call-ID: 6994090221311298789-1168941523@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '7185817751d170532b4bce912c8cd142@217.113.64.11' [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 7185817751d170532b4bce912c8cd142@217.113.64.11 Really destroying SIP dialog '7185817751d170532b4bce912c8cd142@217.113.64.11' Method: INVITE [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18349]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 16 10:59:11] DEBUG[18360]: chan_sip.c:5576 reqprep: Strict routing enforced for session 126a554e5a5c424454784837291c97de@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Reliably Transmitting (no NAT) to 216.226.69.244:5060: INFO sip:2209825337@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK209b342b;rport From: "004769838857" ;tag=as1f5067ce To: ;tag=t1168941520-co1455 Contact: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 103 INFO User-Agent: gatewaycomms Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=# Duration=250 --- [Jan 16 10:59:11] DEBUG[18360]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #274 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432204489001@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298859-1168941550@172.16.100.100 From: ;tag=28651 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d2b-ac106464-3520774 Contact: sip:2425323347@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 226 v=0 o=MG4000|2.0 75602 75602 IN IP4 10.57.2.137 s=- c=IN IP4 10.57.2.137 t=0 0 m=audio 30132 RTP/AVP 18 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=yes a=ptime:40 a=rtpmap:13 CN/8000 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432204489001@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298859-1168941550@172.16.100.100 (54) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=28651 (63) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d2b-ac106464-3 520774 (85) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2425323347@172.16.100.100:5060;user=phone (54) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 226 (19) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 75602 75602 IN IP4 10.57.2.137 (43) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.137 (20) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 30132 RTP/AVP 18 0 101 13 (33) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311298859-1168941550@172.16.100.100 - INVITE (With RT P) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1670 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.100.100 : 5060 (no NAT) Using INVITE request as basis request - 6994090221311298859-1168941550@172.16.100.100 Found peer 'VERAZ' [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 13 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.137:30132 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found description format CN for ID 13 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.57.2.137:30132 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432204489001 in default (domain 10.100.20.11) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2425323347@172.16.100.100:5060;user=phone list_route: hop: [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:13365 handle_request_invite: SIP/5060-09dc5478: New call is still down.... Trying... <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d2b-ac106464-3520774;received=172.16.100.100 From: ;tag=28651 To: Call-ID: 6994090221311298859-1168941550@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:11] DEBUG[18326]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09dc5478 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:11] DEBUG[18375]: pbx.c:1768 pbx_extension_helper: Launching 'Dial' [Jan 16 10:59:11] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: -- Executing [0432204489001@default:1] Dial("SIP/5060-09dc5478", "SIP/2204489001@2 16.226.69.244||t") in new stack Checking device state for peer 5060 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 16 10:59:11] DEBUG[18375]: channel.c:3187 ast_channel_inherit_variables: Not copying variable STACK-default-0432204489001-1. [Jan 16 10:59:11] DEBUG[18375]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 16 10:59:11] DEBUG[18375]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 16 10:59:11] DEBUG[18375]: channel.c:3187 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:2804 sip_call: Outgoing Call for 2204489001 [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 11294 Adding codec 0x100 (g729) to SDP [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2204489001@216.226.69.244 SIP/2.0 (44) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK08767bce;rport (64) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 2: From: "2425323347" ;tag=as21d6ab3a (64) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 7d5db9595d89ebec29c2defd52a709ef@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 9: Date: Tue, 16 Jan 2007 09:59:11 GMT (35) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: o=root 18318 18318 IN IP4 217.113.64.11 (39) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: m=audio 11294 RTP/AVP 18 (24) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2204489001@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK08767bce;rport From: "2425323347" ;tag=as21d6ab3a To: Contact: Call-ID: 7d5db9595d89ebec29c2defd52a709ef@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 11294 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:11] DEBUG[18375]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #275 -- Called 2204489001@216.226.69.244 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1beb9c47 To: ;tag=t1168941539-co1541 From: "934690046" ;tag=as46cc04f6 Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1541-CPO00416 Content-Type: application/sdp Content-Length: 230 v=0 o=- 2075814817 2075814817 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43292 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK1beb9c47 (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941539-co1541 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "934690046" ;tag=as46cc04f6 (62) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1541-CPO00416 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 2075814817 2075814817 IN IP4 216.226.69.244 (47) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 43292 RTP/AVP 18 101 (28) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '15cb97af09d6b84f2345358566576ca3@217.113.64.11' of Request 102: M atch Not Found [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43292 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09db30f0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:43292 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5576 reqprep: Strict routing enforced for session 15cb97af09d6b84f2345358566576ca3@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2205667389@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2beb018a;rport From: "934690046" ;tag=as46cc04f6 To: ;tag=t1168941539-co1541 Contact: Call-ID: 15cb97af09d6b84f2345358566576ca3@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:11] DEBUG[18367]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09db30f0 -- SIP/216.226.69.244-09db30f0 answered SIP/5060-09fb24f8 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:59:11] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:59:11] DEBUG[18367]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09fb24f8 [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-09fb24f8 [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:11] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:11] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 10404 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3520755;received=172.16.100.100 From: ;tag=23199 To: ;tag=as31e8d0e9 Call-ID: 6994090221311298835-1168941538@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18319 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 10404 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:11] DEBUG[18367]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #277 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205667389@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298835-1168941538@172.16.100.100 From: ;tag=23199 To: ;tag=as31e8d0e9 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3520755 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205667389@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298835-1168941538@172.16.100.100 (54) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=23199 (62) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as31e8d0e9 (67) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d13-ac106464-3 520755 (85) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #277 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311298835-1168941538@172.16.100.100' of Response 1: Mat ch Not Found [Jan 16 10:59:11] DEBUG[18367]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5b75ada5 To: ;tag=t1168941547-co1563 From: "4532322555" ;tag=as6d2feea1 Call-ID: 71c97aa5150eb02b7b661cce06bba94e@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1563-CPO00523 Content-Type: application/sdp Content-Length: 228 v=0 o=- 194863634 194863634 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 42640 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5b75ada5 (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941547-co1563 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "4532322555" ;tag=as6d2feea1 (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 71c97aa5150eb02b7b661cce06bba94e@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1563-CPO00523 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: o=- 194863634 194863634 IN IP4 216.226.69.244 (45) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: m=audio 42640 RTP/AVP 18 101 (28) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '71c97aa5150eb02b7b661cc e06bba94e@217.113.64.11' Request 102: Found [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:42640 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-09eb46f0 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.226.69.237:42640 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-09eb46f0 is making progress passing it to SIP/5060-09f01758 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18360]: chan_sip.c:5576 reqprep: Strict routing enforced for session 126a554e5a5c424454784837291c97de@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Reliably Transmitting (no NAT) to 216.226.69.244:5060: INFO sip:2209825337@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5dc15344;rport From: "004769838857" ;tag=as1f5067ce To: ;tag=t1168941520-co1455 Contact: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 104 INFO User-Agent: gatewaycomms Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=# Duration=250 --- [Jan 16 10:59:11] DEBUG[18360]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #278 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18369]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:11] DEBUG[18369]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK641931be To: From: "243851213877" ;tag=as6a3615af Call-ID: 1da2ebd934fc944f608dadd04bc25533@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1584-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK641931be (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "243851213877" ;tag=as6a3615af (68) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 1da2ebd934fc944f608dadd04bc25533@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1584-CPO00000 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #268 - INVITE (got response) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1da2ebd934fc944f608dadd 04bc25533@217.113.64.11' Request 102: Found [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18374]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:59:11] DEBUG[18374]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:11] DEBUG[18374]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:11] DEBUG[18374]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15208 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:11] DEBUG[18374]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:11] DEBUG[18374]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d29-ac106464-3520771;received=172.16.100.100 From: ;tag=28651 To: ;tag=as1aa7d729 Call-ID: 6994090221311298857-1168941550@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15208 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:11] DEBUG[18374]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:11] DEBUG[18374]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 To: ;tag=t1168941548-co1571 From: "972370597" ;tag=as12354d03 Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1571-CPO00554 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941548-co1571 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972370597" ;tag=as12354d03 (62) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1571-CPO00554 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #272 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11' of Request 102: M atch Not Found localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 To: ;tag=t1168941548-co1571 From: "972370597" ;tag=as12354d03 Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1571-CPO00554 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK37cc3953 (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941548-co1571 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "972370597" ;tag=as12354d03 (62) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1571-CPO00554 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11' of Request 102: M atch Found [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2208101110@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK37cc3953;rport From: "972370597" ;tag=as12354d03 To: ;tag=t1168941548-co1571 Contact: Call-ID: 6b34a7636ed1c7bb3a938e2944d80fa4@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #255 (3) INVITE - 5 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #255)) Retransmitting #3 (no NAT) to 196.29.163.6:5060: INVITE sip:249734822305@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK3209708c;rport From: "2356756282" ;tag=as6ddb37ad To: Contact: Call-ID: 001f622a10c230d73b1c9ae457dd3ce6@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 16 Jan 2007 09:59:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 18318 18318 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16866 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK209b342b To: ;tag=t1168941520-co1455 From: "004769838857" ;tag=as1f5067ce Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 103 INFO User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 Content-Length: 0 <-------------> [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK209b342b (64) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941520-co1455 (58) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "004769838857" ;tag=as1f5067ce (68) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 (55) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 103 INFO (14) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 (61) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #274 [Jan 16 10:59:11] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '126a554e5a5c424454784837291c97de@217.113.64.11' of Request 103: M atch Not Found [Jan 16 10:59:11] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK08767bce To: From: "2425323347" ;tag=as21d6ab3a Call-ID: 7d5db9595d89ebec29c2defd52a709ef@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1591-CPO00000 Content-Length: 0 <-------------> [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK08767bce (64) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "2425323347" ;tag=as21d6ab3a (64) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 7d5db9595d89ebec29c2defd52a709ef@217.113.64.11 (55) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO1591-CPO00000 (58) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #275 - INVITE (got response) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7d5db9595d89ebec29c2def d52a709ef@217.113.64.11' Request 102: Found [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18375]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 16 10:59:12] DEBUG[18360]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes [Jan 16 10:59:12] DEBUG[18375]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 16 10:59:12] DEBUG[18375]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 16 10:59:12] DEBUG[18375]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19368 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 10:59:12] DEBUG[18375]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 16 10:59:12] DEBUG[18375]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0d2b-ac106464-3520774;received=172.16.100.100 From: ;tag=28651 To: ;tag=as71709cd6 Call-ID: 6994090221311298859-1168941550@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18318 18318 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19368 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 16 10:59:12] DEBUG[18375]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 16 10:59:12] DEBUG[18375]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> BYE sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311298781-1168941520@172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 CSeq: 4 BYE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520775 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: BYE sip:0432209825337@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311298781-1168941520@172.16.100.100 (54) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=15022 (65) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as42a70485 (67) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: CSeq: 4 BYE (11) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3 520775 (85) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:14502 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 172.16.100.100 : 5060 (no NAT) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311298781-1168941520@172.16.100.100 [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:14079 handle_request_bye: Received bye, issuing owner hangup localhost*CLI> <--- Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-61100000004b0cdd-ac106464-3520775;received=172.16.100.100 From: ;tag=15022 To: ;tag=as42a70485 Call-ID: 6994090221311298781-1168941520@172.16.100.100 CSeq: 4 BYE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 16 10:59:12] DEBUG[18360]: channel.c:3684 ast_generic_bridge: Didn't get a frame from channel: SIP/5060-09d74158 [Jan 16 10:59:12] DEBUG[18360]: channel.c:3988 ast_channel_bridge: Bridge stops bridging channels SIP/5060-09d74158 and SIP/216.226.69.244-09ddd560 [Jan 16 10:59:12] DEBUG[18360]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-09ddd560' [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-09ddd560, SIP callid 126a554e5a5c424454784837291c97de@217.113 .64.11) [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:3278 sip_hangup: update_call_counter(2209825337) - decrement call limit counter on hangup [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call Scheduling destruction of SIP dialog '126a554e5a5c424454784837291c97de@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:5576 reqprep: Strict routing enforced for session 126a554e5a5c424454784837291c97de@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 216.226.69.244, port 5060 Reliably Transmitting (no NAT) to 216.226.69.244:5060: BYE sip:2209825337@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK6c7ec283;rport From: "004769838857" ;tag=as1f5067ce To: ;tag=t1168941520-co1455 Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 105 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #283 [Jan 16 10:59:12] DEBUG[18360]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-09ddd560 [Jan 16 10:59:12] DEBUG[18360]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 16 10:59:12] DEBUG[18360]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 16 10:59:12] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 16 10:59:12] DEBUG[18360]: pbx.c:2364 __ast_pbx_run: Spawn extension (default,0432209825337,1) exited non-zero on 'SIP/5060-09d74158' == Spawn extension (default, 0432209825337, 1) exited non-zero on 'SIP/5060-09d74158' [Jan 16 10:59:12] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 16 10:59:12] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 16 10:59:12] DEBUG[18360]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 16 10:59:12] DEBUG[18360]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-09d74158' [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-09d74158, SIP callid 6994090221311298781-1168941520@172.16.100.100) [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 16 10:59:12] DEBUG[18360]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 16 10:59:12] DEBUG[18360]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-09d74158 [Jan 16 10:59:12] DEBUG[18321]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 16 10:59:12] DEBUG[18321]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 16 10:59:12] DEBUG[18321]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5dc15344 To: ;tag=t1168941520-co1455 From: "004769838857" ;tag=as1f5067ce Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 CSeq: 104 INFO User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 Content-Length: 0 <-------------> [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5dc15344 (64) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168941520-co1455 (58) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 3: From: "004769838857" ;tag=as1f5067ce (68) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 126a554e5a5c424454784837291c97de@217.113.64.11 (55) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 5: CSeq: 104 INFO (14) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1455-CPO00048 (61) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #278 [Jan 16 10:59:12] DEBUG[18326]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '126a554e5a5c424454784837291c97de@217.113.64.11' of Request 104: M atch Not Found Really destroying SIP dialog '6994090221311298781-1168941520@172.16.100.100' Method: BYE [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] DEBUG[18358]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 16 10:59:12] WARNING[18326]: chan_sip.c:1875 retrans_pkt: Maximum retries exceeded on transmission 6994090221311298763-1168941506@172.16.100.100 fo r seqno 1 (Critical Response) Really destroying SIP dialog '6994090221311298763-1168941506@172.16.100.100' Method: CANCEL