[root@localhost bcoppens]# /usr/sbin/asterisk -r Asterisk SVN-branch-1.4-r50468, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk SVN-branch-1.4-r50468 currently running on localhost (pid = 16974) -- Remote UNIX connection [Jan 15 15:35:05] DEBUG[24231]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249912602895@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200803-1168871705@172.16.100.100 From: ;tag=664 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e23-ac106464-3455367 Contact: sip:256755900300@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 220 v=0 o=MG4000|2.0 4091 4091 IN IP4 10.57.2.54 s=- c=IN IP4 10.57.2.54 t=0 0 m=audio 21792 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 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249912602895@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200803-1168871705@172.16.100.100 (54) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=664 (63) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e23-ac106464-3 455367 (85) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:256755900300@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 220 (19) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 4091 4091 IN IP4 10.57.2.54 (40) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.54 (19) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 21792 RTP/AVP 18 101 13 (31) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200803-1168871705@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:05] DEBUG[16996]: 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 - 6994090221311200803-1168871705@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:05] DEBUG[16996]: 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 15 15:35:05] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.54:21792 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 15 15:35:05] DEBUG[16996]: 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 10.57.2.54:21792 [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249912602895 in default (domain 10.100.20.11) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:256755900300@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:05] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08b2b038: 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: Call-ID: 6994090221311200803-1168871705@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 15 15:35:05] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08b2b038 [Jan 15 15:35:05] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:05] DEBUG[24273]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0264249912602895@default:1] Dial("SIP/5060-08b2b038", "SIP/249912602895@196.29.163.6||t") in new stack [Jan 15 15:35:05] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:05] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:05] DEBUG[24273]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:05] DEBUG[24274]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:05] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:05] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "256755900300" ;tag=as63455198 To: CSeq: 102 INVITE Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "256755900300" ;tag=as63455198 (68) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 (55) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport=5060 (69) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10902 - INVITE (got response) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6e294d5a2b73e2fb32333eb f2fdb57a1@217.113.64.11' Request 102: Found [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:06] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[24214]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 15 15:35:06] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[23743]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:06] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200805-1168871706@172.16.100.100 From: ;tag=17502 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e25-ac106464-3455368 Contact: sip:4921199999@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 5422 5422 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 20880 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 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200805-1168871706@172.16.100.100 (54) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=17502 (63) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e25-ac106464-3 455368 (85) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:4921199999@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 5422 5422 IN IP4 10.57.3.22 (40) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 20880 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200805-1168871706@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:06] DEBUG[16996]: 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 - 6994090221311200805-1168871706@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:06] DEBUG[16996]: 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 15 15:35:06] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:20880 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 15 15:35:06] DEBUG[16996]: 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 10.57.3.22:20880 [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205671490 in default (domain 10.100.20.11) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:4921199999@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:06] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08e9f940: 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-6110000000498e25-ac106464-3455368;received=172.16.100.100 From: ;tag=17502 To: Call-ID: 6994090221311200805-1168871706@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 15 15:35:06] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08e9f940 [Jan 15 15:35:06] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:06] DEBUG[24275]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432205671490@default:1] Dial("SIP/5060-08e9f940", "SIP/2205671490@216.226.69.244||t") in new stack [Jan 15 15:35:06] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:06] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:06] DEBUG[24276]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:06] DEBUG[24275]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432205671490-1. [Jan 15 15:35:06] DEBUG[24275]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:06] DEBUG[24275]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:06] DEBUG[24275]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:2804 sip_call: Outgoing Call for 2205671490 [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 11938 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205671490@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK53b1b679;rport (64) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 2: From: "4921199999" ;tag=as0b79991c (64) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 (55) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:06 GMT (35) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: m=audio 11938 RTP/AVP 18 (24) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205671490@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK53b1b679;rport From: "4921199999" ;tag=as0b79991c To: Contact: Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 11938 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:06] DEBUG[24275]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10904 -- Called 2205671490@216.226.69.244 [Jan 15 15:35:06] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:06] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[23923]: 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=z9hG4bK53b1b679 To: From: "4921199999" ;tag=as0b79991c Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO124-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK53b1b679 (64) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0b79991c (64) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 (55) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO124-CPO00000 (57) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10904 - INVITE (got response) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4ee9e3b41426997969f99ac 8669e8665@217.113.64.11' Request 102: Found [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:07] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[24275]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:07] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[24275]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:07] DEBUG[24275]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:07] DEBUG[24275]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 18732 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:07] DEBUG[24275]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:07] DEBUG[24275]: 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-6110000000498e25-ac106464-3455368;received=172.16.100.100 From: ;tag=17502 To: ;tag=as5b835b73 Call-ID: 6994090221311200805-1168871706@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 18732 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 15 15:35:07] DEBUG[24275]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:07] DEBUG[24275]: 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/2.0 200 OK Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK0ea635d1 To: ;tag=t1168871687-co1923 From: "anonymous" ;tag=as4609f0c2 Call-ID: 1ad6a4795e2c16742c0858e27598cab4@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1923-CPO01306 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1650037032 1650037032 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 50640 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 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK0ea635d1 (64) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871687-co1923 (58) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "anonymous" ;tag=as4609f0c2 (62) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 1ad6a4795e2c16742c0858e27598cab4@217.113.64.11 (55) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1923-CPO01306 (61) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1650037032 1650037032 IN IP4 216.226.69.244 (47) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 50640 RTP/AVP 18 101 (28) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '1ad6a4795e2c16742c0858e27598cab4@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:50640 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0876e660 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 193.19.106.196:50640 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 1ad6a4795e2c16742c0858e27598cab4@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:2209993111@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7d94d0d1;rport From: "anonymous" ;tag=as4609f0c2 To: ;tag=t1168871687-co1923 Contact: Call-ID: 1ad6a4795e2c16742c0858e27598cab4@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:07] DEBUG[24210]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0876e660 -- SIP/216.226.69.244-0876e660 answered SIP/anonymous.invalid-08810000 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:07] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:07] DEBUG[24210]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/an onymous.invalid-08810000 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:07] DEBUG[24210]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/anonymous.invalid-08810000 [Jan 15 15:35:07] DEBUG[24277]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - anonymous.invalid [Jan 15 15:35:07] DEBUG[24210]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:07] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer anonymous.invalid [Jan 15 15:35:07] DEBUG[24210]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:07] DEBUG[24210]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11148 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:07] DEBUG[24210]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:07] DEBUG[24210]: 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-6110000000498e07-ac106464-3455350;received=172.16.100.100 From: anonymous ;tag=25255 To: ;tag=as156920e5 Call-ID: 6994090221311200775-1168871687@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 16974 16975 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11148 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 15 15:35:07] DEBUG[24210]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10907 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/anonymous.invalid - state 4 (Invalid) [Jan 15 15:35:07] DEBUG[24278]: app_queue.c:546 changethread: Device 'SIP/anonymous.invalid' changed to state '4' (Invalid) but we don't care because th ey're not a member of any queue. <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432209993111@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200775-1168871687@172.16.100.100 From: anonymous ;tag=25255 To: ;tag=as156920e5 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e07-ac106464-3455350 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432209993111@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200775-1168871687@172.16.100.100 (54) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: anonymous ;tag=25255 (59) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as156920e5 (67) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e07-ac106464-3 455350 (85) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10907 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200775-1168871687@172.16.100.100' of Response 1: Mat ch Not Found [Jan 15 15:35:07] DEBUG[24210]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:07] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 180 Ringing From: "24107371228" ;tag=as3c441b01 To: ;tag=26681538 CSeq: 102 INVITE Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK79f70944;rport=5060 Contact: Content-Length: 151 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 10560228 10560228 IN IP4 10.10.13.4 s=Sip Call c=IN IP4 196.29.163.6 t=0 0 m=audio 40626 RTP/AVP 18 a=rtpmap:18 G729/8000 <-------------> [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "24107371228" ;tag=as3c441b01 (66) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=26681538 (48) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 (55) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK79f70944;rport=5060 (69) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (56) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 151 (19) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=HuaweiSoftX3000 10560228 10560228 IN IP4 10.10.13.4 (53) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=Sip Call (10) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 196.29.163.6 (21) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40626 RTP/AVP 18 (24) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) --- (9 headers 7 lines) --- [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '22580e741be995b944ab373 a6a35e55a@217.113.64.11' Request 102: Found [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 180 to standard invite [Jan 15 15:35:07] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-08a029c8 Found RTP audio format 18 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 15 15:35:07] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 196.29.163.6:40626 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) Found description format G729 for ID 18 [Jan 15 15:35:07] DEBUG[24279]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/196.29.163.6-08a029c8 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 196.29.163.6:40626 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/196.29.163.6-08a029c8 is ringing <--- 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-6110000000498e1f-ac106464-3455365;received=172.16.100.100 From: ;tag=32523 To: ;tag=as65d195fd Call-ID: 6994090221311200799-1168871703@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/196.29.163.6-08a029c8 is making progress passing it to SIP/5060-08586e70 [Jan 15 15:35:07] DEBUG[24263]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:07] DEBUG[24263]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:07] DEBUG[24263]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11890 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:07] DEBUG[24263]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:07] DEBUG[24263]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- 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-6110000000498e1f-ac106464-3455365;received=172.16.100.100 From: ;tag=32523 To: ;tag=as65d195fd Call-ID: 6994090221311200799-1168871703@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11890 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 15 15:35:07] DEBUG[24263]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:07] DEBUG[24263]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432204391307@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200807-1168871707@172.16.100.100 From: ;tag=1573 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e27-ac106464-3455369 Contact: sip:441213288847@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 4185 4185 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 53668 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 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432204391307@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200807-1168871707@172.16.100.100 (54) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1573 (64) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e27-ac106464-3 455369 (85) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:441213288847@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 4185 4185 IN IP4 10.57.3.22 (40) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 53668 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200807-1168871707@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:07] DEBUG[16996]: 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 - 6994090221311200807-1168871707@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:07] DEBUG[16996]: 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 15 15:35:07] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:53668 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 15 15:35:07] DEBUG[16996]: 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 10.57.3.22:53668 [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432204391307 in default (domain 10.100.20.11) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:441213288847@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:07] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08b6fc60: 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-6110000000498e27-ac106464-3455369;received=172.16.100.100 From: ;tag=1573 To: Call-ID: 6994090221311200807-1168871707@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 15 15:35:07] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08b6fc60 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:07] DEBUG[24280]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432204391307@default:1] Dial("SIP/5060-08b6fc60", "SIP/2204391307@216.226.69.244||t") in new stack [Jan 15 15:35:07] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:07] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:07] DEBUG[24281]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:07] DEBUG[24280]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432204391307-1. [Jan 15 15:35:07] DEBUG[24280]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:07] DEBUG[24280]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:07] DEBUG[24280]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:2804 sip_call: Outgoing Call for 2204391307 [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 13164 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2204391307@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK43a6488b;rport (64) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 2: From: "441213288847" ;tag=as735c10a2 (68) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 (55) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:07 GMT (35) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: m=audio 13164 RTP/AVP 18 (24) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2204391307@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK43a6488b;rport From: "441213288847" ;tag=as735c10a2 To: Contact: Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 13164 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:07] DEBUG[24280]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10910 -- Called 2204391307@216.226.69.244 [Jan 15 15:35:07] DEBUG[24275]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:07] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:07] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK53b1b679 To: ;tag=t1168871707-co124 From: "4921199999" ;tag=as0b79991c Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO124-CPO01807 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1081888776 1081888776 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40908 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 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK53b1b679 (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871707-co124 (57) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0b79991c (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO124-CPO01807 (60) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1081888776 1081888776 IN IP4 216.226.69.244 (47) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40908 RTP/AVP 18 101 (28) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4ee9e3b41426997969f99ac 8669e8665@217.113.64.11' Request 102: Found [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40908 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08ea38a8 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 216.226.69.237:40908 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-08ea38a8 is making progress passing it to SIP/5060-08e9f940 [Jan 15 15:35:08] DEBUG[24275]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:08] DEBUG[24275]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:08] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:08] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:08] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:08] DEBUG[24263]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:08] DEBUG[24263]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:08] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:08] DEBUG[24231]: 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=z9hG4bK43a6488b To: From: "441213288847" ;tag=as735c10a2 Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO128-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK43a6488b (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "441213288847" ;tag=as735c10a2 (68) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO128-CPO00000 (57) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10910 - INVITE (got response) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0ee407ed3ebf5f8e0137b11 d4db58b5a@217.113.64.11' Request 102: Found [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:08] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:08] DEBUG[24280]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:08] DEBUG[24280]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:08] DEBUG[24280]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11842 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:08] DEBUG[24280]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:08] DEBUG[24280]: 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-6110000000498e27-ac106464-3455369;received=172.16.100.100 From: ;tag=1573 To: ;tag=as28e54cfd Call-ID: 6994090221311200807-1168871707@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11842 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 15 15:35:08] DEBUG[24280]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:08] DEBUG[24280]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0264249912602895@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200803-1168871705@172.16.100.100 From: ;tag=664 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e23-ac106464-3455367 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0264249912602895@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200803-1168871705@172.16.100.100 (54) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=664 (63) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e23-ac106464-3 455367 (85) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311200803-1168871705@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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@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 15 15:35:08] DEBUG[16996]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10914 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@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 15 15:35:08] DEBUG[24273]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:08] DEBUG[24273]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-08c1db40' [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-08c1db40, SIP callid 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.6 4.11) [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:3278 sip_hangup: update_call_counter(249912602895) - decrement call limit counter on hangup [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 196.29.163.6:5060: CANCEL sip:249912602895@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport From: "256755900300" ;tag=as63455198 To: Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10916 Scheduling destruction of SIP dialog '6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:08] DEBUG[24273]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-08c1db40 [Jan 15 15:35:08] DEBUG[24273]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 15 15:35:08] DEBUG[24273]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,0264249912602895,1) exited non-zero on 'SIP/5060-08b2b038' == Spawn extension (default, 0264249912602895, 1) exited non-zero on 'SIP/5060-08b2b038' [Jan 15 15:35:08] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 6 (Ringing) [Jan 15 15:35:08] DEBUG[24273]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:08] DEBUG[24282]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jan 15 15:35:08] DEBUG[24273]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08b2b038' [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08b2b038, SIP callid 6994090221311200803-1168871705@172.16.100.100) [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:08] DEBUG[24273]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 15 15:35:08] DEBUG[24273]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08b2b038 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:08] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:08] DEBUG[24283]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:08] DEBUG[24161]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:08] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:08] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:08] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:08] DEBUG[24163]: 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=z9hG4bK3fb6e141 To: ;tag=t1168871690-co1952 From: "22993817580" ;tag=as2521d47b Call-ID: 02beef610d69fc533d7a510543bf8595@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1952-CPO01385 Content-Type: application/sdp Content-Length: 230 v=0 o=- 2046299165 2046299165 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 41620 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 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK3fb6e141 (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871690-co1952 (58) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "22993817580" ;tag=as2521d47b (66) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 02beef610d69fc533d7a510543bf8595@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO1952-CPO01385 (61) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 2046299165 2046299165 IN IP4 216.226.69.244 (47) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 41620 RTP/AVP 18 101 (28) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '02beef610d69fc533d7a510543bf8595@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:41620 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0875aa50 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 216.226.69.237:41620 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 02beef610d69fc533d7a510543bf8595@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:2207217670@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK1548751e;rport From: "22993817580" ;tag=as2521d47b To: ;tag=t1168871690-co1952 Contact: Call-ID: 02beef610d69fc533d7a510543bf8595@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:08] DEBUG[24224]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0875aa50 -- SIP/216.226.69.244-0875aa50 answered SIP/5060-088438c0 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:08] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:08] DEBUG[24224]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-088438c0 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:08] DEBUG[24224]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-088438c0 [Jan 15 15:35:08] DEBUG[24284]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:08] DEBUG[24224]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:08] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:08] DEBUG[24224]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:08] DEBUG[24224]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 17582 [Jan 15 15:35:08] DEBUG[24285]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:08] DEBUG[24224]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:08] DEBUG[24224]: 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-6110000000498e0f-ac106464-3455354;received=172.16.100.100 From: ;tag=10233 To: ;tag=as326d6b17 Call-ID: 6994090221311200783-1168871690@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 16974 16975 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 17582 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 15 15:35:08] DEBUG[24224]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10918 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432207217670@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200783-1168871690@172.16.100.100 From: ;tag=10233 To: ;tag=as326d6b17 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e0f-ac106464-3455354 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432207217670@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200783-1168871690@172.16.100.100 (54) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=10233 (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as326d6b17 (67) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e0f-ac106464-3 455354 (85) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10918 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200783-1168871690@172.16.100.100' of Response 1: Mat ch Not Found [Jan 15 15:35:08] DEBUG[24224]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:08] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 200 OK From: "256755900300" ;tag=as63455198 To: CSeq: 102 CANCEL Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "256755900300" ;tag=as63455198 (68) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 CANCEL (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport=5060 (69) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10916 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11' of Request 102: M atch Not Found <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 487 Request Terminated From: "256755900300" ;tag=as63455198 To: ;tag=b3939917 CSeq: 102 INVITE Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "256755900300" ;tag=as63455198 (68) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=b3939917 (48) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport=5060 (69) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11' of Request 102: M atch Found [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249912602895@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK57456829;rport From: "256755900300" ;tag=as63455198 To: ;tag=b3939917 Contact: Call-ID: 6e294d5a2b73e2fb32333ebf2fdb57a1@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432204391025@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200813-1168871708@172.16.100.100 From: ;tag=18411 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e2d-ac106464-3455372 Contact: sip:4921199999@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 3149 3149 IN IP4 10.57.3.21 s=- c=IN IP4 10.57.3.21 t=0 0 m=audio 23976 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 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432204391025@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200813-1168871708@172.16.100.100 (54) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=18411 (63) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e2d-ac106464-3 455372 (85) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:4921199999@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3149 3149 IN IP4 10.57.3.21 (40) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.21 (19) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 23976 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200813-1168871708@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:08] DEBUG[16996]: 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 - 6994090221311200813-1168871708@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:08] DEBUG[16996]: 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 15 15:35:08] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.21:23976 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 15 15:35:08] DEBUG[16996]: 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 10.57.3.21:23976 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432204391025 in default (domain 10.100.20.11) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:4921199999@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-085fd4a8: 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-6110000000498e2d-ac106464-3455372;received=172.16.100.100 From: ;tag=18411 To: Call-ID: 6994090221311200813-1168871708@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 15 15:35:08] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-085fd4a8 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:08] DEBUG[24286]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432204391025@default:1] Dial("SIP/5060-085fd4a8", "SIP/2204391025@216.226.69.244||t") in new stack [Jan 15 15:35:08] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:08] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:08] DEBUG[24287]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:08] DEBUG[24286]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432204391025-1. [Jan 15 15:35:08] DEBUG[24286]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:08] DEBUG[24286]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:08] DEBUG[24286]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:2804 sip_call: Outgoing Call for 2204391025 [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 10022 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2204391025@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK06ef1940;rport (64) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 2: From: "4921199999" ;tag=as4408c867 (64) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 351241e5673707673acf976918fa4282@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:08 GMT (35) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: m=audio 10022 RTP/AVP 18 (24) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2204391025@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK06ef1940;rport From: "4921199999" ;tag=as4408c867 To: Contact: Call-ID: 351241e5673707673acf976918fa4282@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 10022 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:08] DEBUG[24286]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10919 -- Called 2204391025@216.226.69.244 [Jan 15 15:35:08] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- 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=z9hG4bK43a6488b To: ;tag=t1168871707-co128 From: "441213288847" ;tag=as735c10a2 Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO128-CPO01818 Content-Type: application/sdp Content-Length: 228 v=0 o=- 717968811 717968811 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43412 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 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK43a6488b (64) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871707-co128 (57) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "441213288847" ;tag=as735c10a2 (68) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 (55) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO128-CPO01818 (60) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 717968811 717968811 IN IP4 216.226.69.244 (45) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 43412 RTP/AVP 18 101 (28) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0ee407ed3ebf5f8e0137b11 d4db58b5a@217.113.64.11' Request 102: Found [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43412 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-085f9540 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 216.226.69.237:43412 [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:08] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-085f9540 is making progress passing it to SIP/5060-08b6fc60 [Jan 15 15:35:08] DEBUG[24280]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:08] DEBUG[24280]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:08] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:08] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #10914 (1) SIP/2.0 - 1 [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #10914)) 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- 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=z9hG4bK06ef1940 To: From: "4921199999" ;tag=as4408c867 Call-ID: 351241e5673707673acf976918fa4282@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO132-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK06ef1940 (64) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as4408c867 (64) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 351241e5673707673acf976918fa4282@217.113.64.11 (55) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO132-CPO00000 (57) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10919 - INVITE (got response) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '351241e5673707673acf976 918fa4282@217.113.64.11' Request 102: Found [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:09] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:09] DEBUG[24286]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:09] DEBUG[24286]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:09] DEBUG[24286]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19990 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:09] DEBUG[24286]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:09] DEBUG[24286]: 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-6110000000498e2d-ac106464-3455372;received=172.16.100.100 From: ;tag=18411 To: ;tag=as60f84b3a Call-ID: 6994090221311200813-1168871708@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19990 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 15 15:35:09] DEBUG[24286]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:09] DEBUG[24286]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:09] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:09] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:09] DEBUG[22604]: 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=z9hG4bK06ef1940 To: ;tag=t1168871709-co132 From: "4921199999" ;tag=as4408c867 Call-ID: 351241e5673707673acf976918fa4282@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO132-CPO01835 Content-Type: application/sdp Content-Length: 230 v=0 o=- 2143492255 2143492255 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40776 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 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK06ef1940 (64) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871709-co132 (57) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as4408c867 (64) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 351241e5673707673acf976918fa4282@217.113.64.11 (55) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO132-CPO01835 (60) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 2143492255 2143492255 IN IP4 216.226.69.244 (47) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40776 RTP/AVP 18 101 (28) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '351241e5673707673acf976 918fa4282@217.113.64.11' Request 102: Found [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40776 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-085fea28 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 216.226.69.237:40776 [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:09] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-085fea28 is making progress passing it to SIP/5060-085fd4a8 [Jan 15 15:35:09] DEBUG[24286]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:09] DEBUG[24286]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:09] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:10] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:10] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:10] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:10] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #10914 (2) SIP/2.0 - 1 [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #10914)) 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@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 15 15:35:10] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:10] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 180 Ringing From: "24107371228" ;tag=as3c441b01 To: ;tag=26681538 CSeq: 102 INVITE Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK79f70944;rport=5060 Contact: Content-Length: 151 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 10560228 10560229 IN IP4 10.10.13.4 s=Sip Call c=IN IP4 196.29.163.6 t=0 0 m=audio 40626 RTP/AVP 18 a=rtpmap:18 G729/8000 <-------------> [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "24107371228" ;tag=as3c441b01 (66) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=26681538 (48) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 (55) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK79f70944;rport=5060 (69) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (56) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 151 (19) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=HuaweiSoftX3000 10560228 10560229 IN IP4 10.10.13.4 (53) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=Sip Call (10) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 196.29.163.6 (21) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40626 RTP/AVP 18 (24) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) --- (9 headers 7 lines) --- [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '22580e741be995b944ab373 a6a35e55a@217.113.64.11' Request 102: Found [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 180 to standard invite Found RTP audio format 18 [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 196.29.163.6:40626 Found description format G729 for ID 18 [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/196.29.163.6-08a029c8 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 196.29.163.6:40626 [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/196.29.163.6-08a029c8 is ringing -- SIP/196.29.163.6-08a029c8 is making progress passing it to SIP/5060-08586e70 [Jan 15 15:35:10] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432208901690@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200821-1168871710@172.16.100.100 From: ;tag=19320 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e35-ac106464-3455375 Contact: sip:972320621@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 2837 2837 IN IP4 10.57.3.26 s=- c=IN IP4 10.57.3.26 t=0 0 m=audio 58076 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 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432208901690@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200821-1168871710@172.16.100.100 (54) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=19320 (62) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e35-ac106464-3 455375 (85) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:972320621@172.16.100.100:5060;user=phone (53) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 2837 2837 IN IP4 10.57.3.26 (40) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.26 (19) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 58076 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:10] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200821-1168871710@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:10] DEBUG[16996]: 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 - 6994090221311200821-1168871710@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:10] DEBUG[16996]: 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 15 15:35:10] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.26:58076 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 15 15:35:10] DEBUG[16996]: 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 10.57.3.26:58076 [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432208901690 in default (domain 10.100.20.11) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:972320621@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:10] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08e5cf60: 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-6110000000498e35-ac106464-3455375;received=172.16.100.100 From: ;tag=19320 To: Call-ID: 6994090221311200821-1168871710@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 15 15:35:10] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08e5cf60 [Jan 15 15:35:10] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:10] DEBUG[24288]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432208901690@default:1] Dial("SIP/5060-08e5cf60", "SIP/2208901690@216.226.69.244||t") in new stack [Jan 15 15:35:10] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:10] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:10] DEBUG[24289]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:10] DEBUG[24288]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432208901690-1. [Jan 15 15:35:10] DEBUG[24288]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:10] DEBUG[24288]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:10] DEBUG[24288]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:2804 sip_call: Outgoing Call for 2208901690 [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 10010 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2208901690@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7e1fafe7;rport (64) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 2: From: "972320621" ;tag=as506570ea (62) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 (55) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:10 GMT (35) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: m=audio 10010 RTP/AVP 18 (24) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2208901690@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7e1fafe7;rport From: "972320621" ;tag=as506570ea To: Contact: Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 10010 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:10] DEBUG[24288]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10924 -- Called 2208901690@216.226.69.244 [Jan 15 15:35:10] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:10] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:10] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:10] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:10] DEBUG[23743]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:11] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 10.100.20.193:5060 ---> REGISTER sip:10.100.20.11 SIP/2.0 Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK46ae9bfe680505B1 From: "2004" ;tag=1FD02568-93601AB1 To: CSeq: 641 REGISTER Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Authorization: Digest username="2004", realm="gatewaycomms", nonce="78eb6ed6", uri="sip:10.100.20.11", response="7142013c9828701224db195c946e9e83", algo rithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: REGISTER sip:10.100.20.11 SIP/2.0 (33) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK46ae9bfe680505B1 (61) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: "2004" ;tag=1FD02568-93601AB1 (58) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (27) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 641 REGISTER (18) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 (49) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, I NFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (133) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 (49) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Authorization: Digest username="2004", realm="gatewaycomms", nonce="78eb6ed6", uri="sip:10.100.20.11", response="7142013c9828701224db195c946e9e83", algorithm=MD5 (161) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Expires: 3600 (13) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 0 (17) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 - REGISTER (No RTP) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 10.100.20.193 : 5060 (no NAT) localhost*CLI> <--- Transmitting (no NAT) to 10.100.20.193:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK46ae9bfe680505B1;received=10.100.20.193 From: "2004" ;tag=1FD02568-93601AB1 To: Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 CSeq: 641 REGISTER User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> <--- Transmitting (no NAT) to 10.100.20.193:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK46ae9bfe680505B1;received=10.100.20.193 From: "2004" ;tag=1FD02568-93601AB1 To: ;tag=as25326a85 Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 CSeq: 641 REGISTER User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="gatewaycomms", nonce="3c71528e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3f6a5324-5b8864fe-2b203bdf@10.100.20.193' in 32000 ms (Method: REGISTER) [Jan 15 15:35:11] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:11] DEBUG[24161]: 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=z9hG4bK7e1fafe7 To: From: "972320621" ;tag=as506570ea Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO143-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7e1fafe7 (64) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972320621" ;tag=as506570ea (62) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 (55) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO143-CPO00000 (57) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10924 - INVITE (got response) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '62ed0c706f8b2b2d07a2e30 658a0b16b@217.113.64.11' Request 102: Found [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:11] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:11] DEBUG[24288]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:11] DEBUG[24288]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:11] DEBUG[24288]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15888 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:11] DEBUG[24288]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:11] DEBUG[24288]: 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-6110000000498e35-ac106464-3455375;received=172.16.100.100 From: ;tag=19320 To: ;tag=as614a422b Call-ID: 6994090221311200821-1168871710@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15888 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 15 15:35:11] DEBUG[24288]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:11] DEBUG[24288]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 localhost*CLI> <--- SIP read from 10.100.20.193:5060 ---> REGISTER sip:10.100.20.11 SIP/2.0 Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK2571b46891FFB7D3 From: "2004" ;tag=1FD02568-93601AB1 To: CSeq: 642 REGISTER Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Authorization: Digest username="2004", realm="gatewaycomms", nonce="3c71528e", uri="sip:10.100.20.11", response="4090dbd91ad089137831897e9c8f0a70", algo rithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: REGISTER sip:10.100.20.11 SIP/2.0 (33) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK2571b46891FFB7D3 (61) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: "2004" ;tag=1FD02568-93601AB1 (58) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (27) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 642 REGISTER (18) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 (49) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, I NFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (133) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 (49) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Authorization: Digest username="2004", realm="gatewaycomms", nonce="3c71528e", uri="sip:10.100.20.11", response="4090dbd91ad089137831897e9c8f0a70", algorithm=MD5 (161) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Expires: 3600 (13) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 0 (17) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 10.100.20.193 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.100.20.193:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK2571b46891FFB7D3;received=10.100.20.193 From: "2004" ;tag=1FD02568-93601AB1 To: Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 CSeq: 642 REGISTER User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.100.20.193:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.20.193;branch=z9hG4bK2571b46891FFB7D3;received=10.100.20.193 From: "2004" ;tag=1FD02568-93601AB1 To: ;tag=as25326a85 Call-ID: 3f6a5324-5b8864fe-2b203bdf@10.100.20.193 CSeq: 642 REGISTER User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Mon, 15 Jan 2007 14:35:11 GMT Content-Length: 0 <------------> [Jan 15 15:35:11] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/20 04 Scheduling destruction of SIP dialog '3f6a5324-5b8864fe-2b203bdf@10.100.20.193' in 32000 ms (Method: REGISTER) [Jan 15 15:35:11] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2004 [Jan 15 15:35:11] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 2004 [Jan 15 15:35:11] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/2004 - state 1 (Not in use) [Jan 15 15:35:11] DEBUG[24290]: app_queue.c:546 changethread: Device 'SIP/2004' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 15 15:35:11] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:11] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:11] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[23743]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249126780567@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200823-1168871711@172.16.100.100 From: ;tag=3390 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e37-ac106464-3455376 Contact: sip:2356281281@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 9503 9503 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 20216 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 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249126780567@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200823-1168871711@172.16.100.100 (54) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=3390 (62) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e37-ac106464-3 455376 (85) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356281281@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9503 9503 IN IP4 10.57.2.121 (41) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 20216 RTP/AVP 18 101 13 (31) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200823-1168871711@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:11] DEBUG[16996]: 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 - 6994090221311200823-1168871711@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:11] DEBUG[16996]: 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 15 15:35:11] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:20216 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 15 15:35:11] DEBUG[16996]: 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 10.57.2.121:20216 [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249126780567 in default (domain 10.100.20.11) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356281281@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08a944e8: 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-6110000000498e37-ac106464-3455376;received=172.16.100.100 From: ;tag=3390 To: Call-ID: 6994090221311200823-1168871711@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 15 15:35:11] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08a944e8 [Jan 15 15:35:11] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:11] DEBUG[24291]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0264249126780567@default:1] Dial("SIP/5060-08a944e8", "SIP/249126780567@196.29.163.6||t") in new stack [Jan 15 15:35:11] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:11] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:11] DEBUG[24292]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:11] DEBUG[24291]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0264249126780567-1. [Jan 15 15:35:11] DEBUG[24291]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:11] DEBUG[24291]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:11] DEBUG[24291]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:2804 sip_call: Outgoing Call for 249126780567 [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 14814 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249126780567@196.29.163.6 SIP/2.0 (44) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport (64) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 2: From: "2356281281" ;tag=as67191451 (64) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 (55) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:11 GMT (35) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: m=audio 14814 RTP/AVP 18 (24) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249126780567@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport From: "2356281281" ;tag=as67191451 To: Contact: Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 14814 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:11] DEBUG[24291]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10930 -- Called 249126780567@196.29.163.6 [Jan 15 15:35:11] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:11] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- 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=z9hG4bK7e1fafe7 To: ;tag=t1168871710-co143 From: "972320621" ;tag=as506570ea Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO143-CPO01873 Content-Type: application/sdp Content-Length: 228 v=0 o=- 750509029 750509029 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40132 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 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7e1fafe7 (64) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871710-co143 (57) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972320621" ;tag=as506570ea (62) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 (55) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO143-CPO01873 (60) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 750509029 750509029 IN IP4 216.226.69.244 (45) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40132 RTP/AVP 18 101 (28) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '62ed0c706f8b2b2d07a2e30 658a0b16b@217.113.64.11' Request 102: Found [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40132 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08801420 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 216.226.69.237:40132 [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-08801420 is making progress passing it to SIP/5060-08e5cf60 [Jan 15 15:35:11] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:11] DEBUG[24288]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:11] DEBUG[24288]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:11] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:11] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432207728414@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200825-1168871711@172.16.100.100 From: ;tag=3390 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e39-ac106464-3455377 Contact: sip:972641064@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 3184 3184 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 42032 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 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432207728414@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200825-1168871711@172.16.100.100 (54) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=3390 (61) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e39-ac106464-3 455377 (85) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:972641064@172.16.100.100:5060;user=phone (53) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3184 3184 IN IP4 10.57.3.22 (40) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 42032 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200825-1168871711@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:11] DEBUG[16996]: 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 - 6994090221311200825-1168871711@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:11] DEBUG[16996]: 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 15 15:35:11] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:42032 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 15 15:35:11] DEBUG[16996]: 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 10.57.3.22:42032 [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432207728414 in default (domain 10.100.20.11) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:972641064@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-0868e648: 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-6110000000498e39-ac106464-3455377;received=172.16.100.100 From: ;tag=3390 To: Call-ID: 6994090221311200825-1168871711@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 15 15:35:11] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0868e648 [Jan 15 15:35:11] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:11] DEBUG[24293]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432207728414@default:1] Dial("SIP/5060-0868e648", "SIP/2207728414@216.226.69.244||t") in new stack [Jan 15 15:35:11] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:11] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:11] DEBUG[24294]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:11] DEBUG[24293]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432207728414-1. [Jan 15 15:35:11] DEBUG[24293]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:11] DEBUG[24293]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:11] DEBUG[24293]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:2804 sip_call: Outgoing Call for 2207728414 [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 16178 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2207728414@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5206503b;rport (64) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 2: From: "972641064" ;tag=as1883046c (62) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 4: Contact: (38) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 (55) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:11 GMT (35) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: m=audio 16178 RTP/AVP 18 (24) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2207728414@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5206503b;rport From: "972641064" ;tag=as1883046c To: Contact: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 16178 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:11] DEBUG[24293]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10933 -- Called 2207728414@216.226.69.244 [Jan 15 15:35:11] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356281281" ;tag=as67191451 To: CSeq: 102 INVITE Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "2356281281" ;tag=as67191451 (64) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 (55) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport=5060 (69) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:11] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) [Jan 15 15:35:11] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #10914 (3) SIP/2.0 - 1 [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #10914)) 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@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 15 15:35:12] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[23844]: 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=z9hG4bK5206503b To: From: "972641064" ;tag=as1883046c Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO160-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5206503b (64) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972641064" ;tag=as1883046c (62) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 (55) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO160-CPO00000 (57) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10933 - INVITE (got response) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '796ca1b868f7cce6709aafe c493b6158@217.113.64.11' Request 102: Found [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:12] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[24275]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[24275]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK53b1b679 To: ;tag=t1168871707-co124 From: "4921199999" ;tag=as0b79991c Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 CSeq: 102 INVITE Reason: Q.850 ;cause=41 ;text="Unknown" Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO124-CPO01807 Content-Length: 0 <-------------> [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 503 Service Unavailable (31) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK53b1b679 (64) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871707-co124 (57) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0b79991c (64) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 (55) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850 ;cause=41 ;text="Unknown" (39) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: (45) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO124-CPO01807 (60) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 0 (17) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:14689 sipsock_read: Failed to grab owner channel lock, trying again. (SIP call 4ee9e3b41426997969f99ac8669e86 65@217.113.64.11) [Jan 15 15:35:12] DEBUG[24293]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:12] DEBUG[24293]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:12] DEBUG[24293]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:12] DEBUG[24293]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 17776 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:12] DEBUG[24293]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:12] DEBUG[24293]: 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-6110000000498e39-ac106464-3455377;received=172.16.100.100 From: ;tag=3390 To: ;tag=as191dc5fc Call-ID: 6994090221311200825-1168871711@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 17776 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 15 15:35:12] DEBUG[24293]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:12] DEBUG[24293]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '4ee9e3b41426997969f99ac8669e8665@217.113.64.11' of Request 102: M atch Not Found -- Got SIP response 503 "Service Unavailable" back from 216.226.69.244 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2205671490@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK53b1b679;rport From: "4921199999" ;tag=as0b79991c To: ;tag=t1168871707-co124 Contact: Call-ID: 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 4ee9e3b41426997969f99ac8669e8665@217.113.64.11 -- SIP/216.226.69.244-08ea38a8 is circuit-busy [Jan 15 15:35:12] DEBUG[24275]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-08ea38a8' [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-08ea38a8, SIP callid 4ee9e3b41426997969f99ac8669e8665@217.113 .64.11) [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:3278 sip_hangup: update_call_counter(2205671490) - decrement call limit counter on hangup [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:12] DEBUG[24275]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-08ea38a8 == Everyone is busy/congested at this time (1:0/1/0) [Jan 15 15:35:12] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:12] DEBUG[24275]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:12] DEBUG[24275]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. [Jan 15 15:35:12] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:12] DEBUG[24275]: pbx.c:1767 pbx_extension_helper: Launching 'Goto' -- Executing [0432205671490@default:2] Goto("SIP/5060-08e9f940", "s-CONGESTION|1") in new stack [Jan 15 15:35:12] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) -- Goto (default,s-CONGESTION,1) [Jan 15 15:35:12] DEBUG[24295]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:12] DEBUG[24275]: pbx.c:1767 pbx_extension_helper: Launching 'Hangup' -- Executing [s-CONGESTION@default:1] Hangup("SIP/5060-08e9f940", "") in new stack [Jan 15 15:35:12] DEBUG[24275]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,s-CONGESTION,1) exited non-zero on 'SIP/5060-08e9f940' == Spawn extension (default, s-CONGESTION, 1) exited non-zero on 'SIP/5060-08e9f940' [Jan 15 15:35:12] DEBUG[24275]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:12] DEBUG[24275]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08e9f940' [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08e9f940, SIP callid 6994090221311200805-1168871706@172.16.100.100) [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:12] DEBUG[24275]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311200805-1168871706@172.16.100.100' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e25-ac106464-3455368;received=172.16.100.100 From: ;tag=17502 To: ;tag=as5b835b73 Call-ID: 6994090221311200805-1168871706@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 15 15:35:12] DEBUG[24275]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10937 [Jan 15 15:35:12] DEBUG[24275]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08e9f940 [Jan 15 15:35:12] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:12] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:12] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:12] DEBUG[24296]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200805-1168871706@172.16.100.100 From: ;tag=17502 To: ;tag=as5b835b73 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e25-ac106464-3455368 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200805-1168871706@172.16.100.100 (54) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=17502 (63) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as5b835b73 (67) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e25-ac106464-3 455368 (85) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10937 [Jan 15 15:35:12] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200805-1168871706@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '4ee9e3b41426997969f99ac8669e8665@217.113.64.11' Method: INVITE [Jan 15 15:35:12] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:12] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:12] DEBUG[24293]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200827-1168871713@172.16.100.100 From: ;tag=4299 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3b-ac106464-3455378 Contact: sip:4921199999@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 5424 5424 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 22516 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200827-1168871713@172.16.100.100 (54) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (62) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3b-ac106464-3 455378 (85) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:4921199999@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 5424 5424 IN IP4 10.57.3.22 (40) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 22516 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200827-1168871713@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:13] DEBUG[16996]: 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 - 6994090221311200827-1168871713@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:22516 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 15 15:35:13] DEBUG[16996]: 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 10.57.3.22:22516 [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205671490 in default (domain 10.100.20.11) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:4921199999@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08ea38a8: 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-6110000000498e3b-ac106464-3455378;received=172.16.100.100 From: ;tag=4299 To: Call-ID: 6994090221311200827-1168871713@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 15 15:35:13] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08ea38a8 [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:13] DEBUG[24297]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432205671490@default:1] Dial("SIP/5060-08ea38a8", "SIP/2205671490@216.226.69.244||t") in new stack [Jan 15 15:35:13] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[24298]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[24297]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432205671490-1. [Jan 15 15:35:13] DEBUG[24297]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:13] DEBUG[24297]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:13] DEBUG[24297]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:2804 sip_call: Outgoing Call for 2205671490 [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 18852 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205671490@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5362c6d6;rport (64) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 2: From: "4921199999" ;tag=as452eaf89 (64) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 (55) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:13 GMT (35) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: m=audio 18852 RTP/AVP 18 (24) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205671490@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5362c6d6;rport From: "4921199999" ;tag=as452eaf89 To: Contact: Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 18852 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10938 -- Called 2205671490@216.226.69.244 [Jan 15 15:35:13] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209927774@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200829-1168871713@172.16.100.100 From: ;tag=4299 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3d-ac106464-3455379 Contact: sip:31643659444@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 1133 1133 IN IP4 10.57.3.26 s=- c=IN IP4 10.57.3.26 t=0 0 m=audio 50612 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209927774@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200829-1168871713@172.16.100.100 (54) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (63) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3d-ac106464-3 455379 (85) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:31643659444@172.16.100.100:5060;user=phone (55) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 1133 1133 IN IP4 10.57.3.26 (40) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.26 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 50612 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200829-1168871713@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:13] DEBUG[16996]: 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 - 6994090221311200829-1168871713@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.26:50612 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 15 15:35:13] DEBUG[16996]: 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 10.57.3.26:50612 [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209927774 in default (domain 10.100.20.11) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:31643659444@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08bd4250: 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-6110000000498e3d-ac106464-3455379;received=172.16.100.100 From: ;tag=4299 To: Call-ID: 6994090221311200829-1168871713@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 15 15:35:13] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08bd4250 [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:13] DEBUG[24299]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' [Jan 15 15:35:13] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: -- Executing [0432209927774@default:1] Dial("SIP/5060-08bd4250", "SIP/2209927774@2 16.226.69.244||t") in new stack Checking device state for peer 5060 [Jan 15 15:35:13] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[24300]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[24299]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432209927774-1. [Jan 15 15:35:13] DEBUG[24299]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:13] DEBUG[24299]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:13] DEBUG[24299]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:2804 sip_call: Outgoing Call for 2209927774 [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 17564 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209927774@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK45c475c6;rport (64) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 2: From: "31643659444" ;tag=as1fa5c7b9 (66) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 4: Contact: (40) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 18cc90fe7207502069e344106bf9f53e@217.113.64.11 (55) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:13 GMT (35) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: m=audio 17564 RTP/AVP 18 (24) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209927774@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK45c475c6;rport From: "31643659444" ;tag=as1fa5c7b9 To: Contact: Call-ID: 18cc90fe7207502069e344106bf9f53e@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 17564 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10940 -- Called 2209927774@216.226.69.244 [Jan 15 15:35:13] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:13] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:13] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:13] DEBUG[24161]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:13] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:13] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[22394]: 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=z9hG4bK5362c6d6 To: From: "4921199999" ;tag=as452eaf89 Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO169-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5362c6d6 (64) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as452eaf89 (64) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 (55) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO169-CPO00000 (57) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10938 - INVITE (got response) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0fe9f566534b27380d95344 c3bfb1d95@217.113.64.11' Request 102: Found [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:13] DEBUG[24297]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 18814 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:13] DEBUG[24297]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:13] DEBUG[24297]: 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-6110000000498e3b-ac106464-3455378;received=172.16.100.100 From: ;tag=4299 To: ;tag=as197d3bd7 Call-ID: 6994090221311200827-1168871713@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 18814 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 15 15:35:13] DEBUG[24297]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:13] DEBUG[24297]: 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=z9hG4bK45c475c6 To: From: "31643659444" ;tag=as1fa5c7b9 Call-ID: 18cc90fe7207502069e344106bf9f53e@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO170-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK45c475c6 (64) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "31643659444" ;tag=as1fa5c7b9 (66) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 18cc90fe7207502069e344106bf9f53e@217.113.64.11 (55) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO170-CPO00000 (57) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10940 - INVITE (got response) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '18cc90fe7207502069e3441 06bf9f53e@217.113.64.11' Request 102: Found [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432204495604@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200831-1168871713@172.16.100.100 From: ;tag=4299 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3f-ac106464-3455380 Contact: sip:4921199999@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 3149 3149 IN IP4 10.57.3.21 s=- c=IN IP4 10.57.3.21 t=0 0 m=audio 32424 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432204495604@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200831-1168871713@172.16.100.100 (54) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (62) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3f-ac106464-3 455380 (85) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:4921199999@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3149 3149 IN IP4 10.57.3.21 (40) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.21 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 32424 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200831-1168871713@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:13] DEBUG[16996]: 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 - 6994090221311200831-1168871713@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.21:32424 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 15 15:35:13] DEBUG[16996]: 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 10.57.3.21:32424 [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432204495604 in default (domain 10.100.20.11) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:4921199999@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08afd2c0: 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-6110000000498e3f-ac106464-3455380;received=172.16.100.100 From: ;tag=4299 To: Call-ID: 6994090221311200831-1168871713@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 localhost*CLI> <------------> [Jan 15 15:35:13] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08afd2c0 [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:13] DEBUG[24301]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432204495604@default:1] Dial("SIP/5060-08afd2c0", "SIP/2204495604@216.226.69.244||t") in new stack [Jan 15 15:35:13] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[24302]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[24301]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432204495604-1. [Jan 15 15:35:13] DEBUG[24301]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:13] DEBUG[24301]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:13] DEBUG[24301]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:2804 sip_call: Outgoing Call for 2204495604 [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 17426 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2204495604@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7de4aefb;rport (64) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 2: From: "4921199999" ;tag=as0028df42 (64) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 (55) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:13 GMT (35) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: m=audio 17426 RTP/AVP 18 (24) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2204495604@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7de4aefb;rport From: "4921199999" ;tag=as0028df42 To: Contact: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 17426 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:13] DEBUG[24301]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10943 -- Called 2204495604@216.226.69.244 [Jan 15 15:35:13] DEBUG[24299]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:13] DEBUG[24297]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 18158 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:13] DEBUG[24299]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- 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-6110000000498e3d-ac106464-3455379;received=172.16.100.100 From: ;tag=4299 To: ;tag=as47a6d354 Call-ID: 6994090221311200829-1168871713@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 18158 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 15 15:35:13] DEBUG[24299]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:13] DEBUG[24299]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:13] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[24299]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:13] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432208100836@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200833-1168871713@172.16.100.100 From: ;tag=4299 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e41-ac106464-3455381 Contact: sip:148708291@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 2103 2103 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 49792 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432208100836@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200833-1168871713@172.16.100.100 (54) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (61) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e41-ac106464-3 455381 (85) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:148708291@172.16.100.100:5060;user=phone (53) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 2103 2103 IN IP4 10.57.3.22 (40) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 49792 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200833-1168871713@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:13] DEBUG[16996]: 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 - 6994090221311200833-1168871713@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:13] DEBUG[16996]: 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 15 15:35:13] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:49792 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 15 15:35:13] DEBUG[16996]: 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 10.57.3.22:49792 [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432208100836 in default (domain 10.100.20.11) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:148708291@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:13] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08b027a8: 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-6110000000498e41-ac106464-3455381;received=172.16.100.100 From: ;tag=4299 To: Call-ID: 6994090221311200833-1168871713@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 15 15:35:13] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08b027a8 [Jan 15 15:35:13] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:13] DEBUG[24303]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432208100836@default:1] Dial("SIP/5060-08b027a8", "SIP/2208100836@216.226.69.244||t") in new stack [Jan 15 15:35:13] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:14] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '1da5a7a65068c2514435c00b0786556d@217.113.64.11' [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 1da5a7a65068c2514435c00b0786556d@217.113.64.11 Really destroying SIP dialog '1da5a7a65068c2514435c00b0786556d@217.113.64.11' Method: INVITE [Jan 15 15:35:14] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:14] DEBUG[24297]: rtp.c:870 ast_rtcp_read: Got RTCP report of 48 bytes [Jan 15 15:35:14] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:14] DEBUG[24299]: 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=z9hG4bK5362c6d6 To: ;tag=t1168871713-co169 From: "4921199999" ;tag=as452eaf89 Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO169-CPO01939 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1589646293 1589646293 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 42136 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 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5362c6d6 (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871713-co169 (57) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as452eaf89 (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 (55) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO169-CPO01939 (60) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1589646293 1589646293 IN IP4 216.226.69.244 (47) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 42136 RTP/AVP 18 101 (28) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0fe9f566534b27380d95344 c3bfb1d95@217.113.64.11' Request 102: Found [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:42136 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08c7db18 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 216.226.69.237:42136 [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:14] DEBUG[24297]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:14] DEBUG[24297]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 -- SIP/216.226.69.244-08c7db18 is making progress passing it to SIP/5060-08ea38a8 [Jan 15 15:35:14] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[24286]: 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=z9hG4bK7de4aefb To: From: "4921199999" ;tag=as0028df42 Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO174-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7de4aefb (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0028df42 (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 (55) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO174-CPO00000 (57) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10943 - INVITE (got response) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '63d06de74913b81e3da871c c28c5a258@217.113.64.11' Request 102: Found [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:14] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:14] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[24301]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:14] DEBUG[24301]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:14] DEBUG[24301]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11236 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:14] DEBUG[24301]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:14] DEBUG[24301]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- 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-6110000000498e3f-ac106464-3455380;received=172.16.100.100 From: ;tag=4299 To: ;tag=as7c2ffa47 Call-ID: 6994090221311200831-1168871713@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11236 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 15 15:35:14] DEBUG[24301]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:14] DEBUG[24301]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:14] DEBUG[23986]: 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=z9hG4bK51c45a90 To: From: "148708291" ;tag=as76d897bc Call-ID: 65b68e416f5d257e122182b77d037372@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO175-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK51c45a90 (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "148708291" ;tag=as76d897bc (62) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 65b68e416f5d257e122182b77d037372@217.113.64.11 (55) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO175-CPO00000 (57) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10946 - INVITE (got response) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '65b68e416f5d257e122182b 77d037372@217.113.64.11' Request 102: Found [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:14] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:14] DEBUG[24303]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:14] DEBUG[24303]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:14] DEBUG[24303]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:14] DEBUG[24303]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 12490 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:14] DEBUG[24303]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:14] DEBUG[24303]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- 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-6110000000498e41-ac106464-3455381;received=172.16.100.100 From: ;tag=4299 To: ;tag=as05b2fe94 Call-ID: 6994090221311200833-1168871713@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 12490 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 15 15:35:14] DEBUG[24303]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:14] DEBUG[24303]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:14] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 404 Not Found From: "2356281281" ;tag=as67191451 To: ;tag=567abb1d CSeq: 102 INVITE Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport=5060 Reason: Q.850;cause=1;text="Unallocated number" Content-Length: 0 <-------------> [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 404 Not Found (21) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "2356281281" ;tag=as67191451 (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=567abb1d (48) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 (55) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport=5060 (69) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850;cause=1;text="Unallocated number" (47) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 404 to standard invite Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249126780567@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7bbb7f44;rport From: "2356281281" ;tag=as67191451 To: ;tag=567abb1d Contact: Call-ID: 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11 -- SIP/196.29.163.6-086867b0 is circuit-busy [Jan 15 15:35:14] DEBUG[24291]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-086867b0' [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-086867b0, SIP callid 2e65e949196ed69a39dc82180a6ab3d4@217.113.6 4.11) [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:3278 sip_hangup: update_call_counter(249126780567) - decrement call limit counter on hangup [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:14] DEBUG[24291]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-086867b0 == Everyone is busy/congested at this time (1:0/1/0) [Jan 15 15:35:14] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 15 15:35:14] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:14] DEBUG[24291]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:14] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 6 (Ringing) [Jan 15 15:35:14] DEBUG[24291]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. [Jan 15 15:35:14] DEBUG[24305]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jan 15 15:35:14] DEBUG[24291]: pbx.c:1767 pbx_extension_helper: Launching 'Goto' -- Executing [0264249126780567@default:2] Goto("SIP/5060-08a944e8", "s-CONGESTION|1") in new stack -- Goto (default,s-CONGESTION,1) [Jan 15 15:35:14] DEBUG[24291]: pbx.c:1767 pbx_extension_helper: Launching 'Hangup' -- Executing [s-CONGESTION@default:1] Hangup("SIP/5060-08a944e8", "") in new stack [Jan 15 15:35:14] DEBUG[24291]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,s-CONGESTION,1) exited non-zero on 'SIP/5060-08a944e8' == Spawn extension (default, s-CONGESTION, 1) exited non-zero on 'SIP/5060-08a944e8' [Jan 15 15:35:14] DEBUG[24291]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:14] DEBUG[24291]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08a944e8' [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08a944e8, SIP callid 6994090221311200823-1168871711@172.16.100.100) [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:14] DEBUG[24291]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311200823-1168871711@172.16.100.100' in 32000 ms (Method: INVITE) <--- 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-6110000000498e37-ac106464-3455376;received=172.16.100.100 From: ;tag=3390 To: ;tag=as6ddb80c1 Call-ID: 6994090221311200823-1168871711@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 15 15:35:14] DEBUG[24291]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10952 [Jan 15 15:35:14] DEBUG[24291]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08a944e8 [Jan 15 15:35:14] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:14] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:14] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:14] DEBUG[24306]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0264249126780567@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200823-1168871711@172.16.100.100 From: ;tag=3390 To: ;tag=as6ddb80c1 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e37-ac106464-3455376 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0264249126780567@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200823-1168871711@172.16.100.100 (54) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=3390 (62) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as6ddb80c1 (70) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e37-ac106464-3 455376 (85) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10952 [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200823-1168871711@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '2e65e949196ed69a39dc82180a6ab3d4@217.113.64.11' Method: INVITE [Jan 15 15:35:14] DEBUG[24161]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:14] DEBUG[24303]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:14] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- 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=z9hG4bK7de4aefb To: ;tag=t1168871714-co174 From: "4921199999" ;tag=as0028df42 Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO174-CPO01951 Content-Type: application/sdp Content-Length: 228 v=0 o=- 790356290 790356290 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40016 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 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7de4aefb (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871714-co174 (57) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0028df42 (64) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 (55) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO174-CPO01951 (60) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 790356290 790356290 IN IP4 216.226.69.244 (45) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40016 RTP/AVP 18 101 (28) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '63d06de74913b81e3da871c c28c5a258@217.113.64.11' Request 102: Found [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40016 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08b01228 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 216.226.69.237:40016 [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:14] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-08b01228 is making progress passing it to SIP/5060-08afd2c0 [Jan 15 15:35:14] DEBUG[24301]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:14] DEBUG[24301]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:14] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:14] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[22636]: 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=z9hG4bK51c45a90 To: ;tag=t1168871714-co175 From: "148708291" ;tag=as76d897bc Call-ID: 65b68e416f5d257e122182b77d037372@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO175-CPO01952 Content-Type: application/sdp Content-Length: 228 v=0 o=- 165000384 165000384 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43604 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 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK51c45a90 (64) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871714-co175 (57) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "148708291" ;tag=as76d897bc (62) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 65b68e416f5d257e122182b77d037372@217.113.64.11 (55) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO175-CPO01952 (60) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 165000384 165000384 IN IP4 216.226.69.244 (45) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 43604 RTP/AVP 18 101 (28) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '65b68e416f5d257e122182b 77d037372@217.113.64.11' Request 102: Found [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43604 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08a42a80 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 216.226.69.237:43604 [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-08a42a80 is making progress passing it to SIP/5060-08b027a8 [Jan 15 15:35:15] DEBUG[24303]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:15] DEBUG[24303]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:15] DEBUG[23743]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> BYE sip:0432209847094@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200461-1168871480@172.16.100.100 From: ;tag=13133 To: ;tag=as4939a3af CSeq: 2 BYE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498ccd-ac106464-3455382 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: BYE sip:0432209847094@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200461-1168871480@172.16.100.100 (54) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=13133 (65) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as4939a3af (67) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 2 BYE (11) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498ccd-ac106464-3 455382 (85) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 172.16.100.100 : 5060 (no NAT) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311200461-1168871480@172.16.100.100 [Jan 15 15:35:15] DEBUG[16996]: 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-6110000000498ccd-ac106464-3455382;received=172.16.100.100 From: ;tag=13133 To: ;tag=as4939a3af Call-ID: 6994090221311200461-1168871480@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 15 15:35:15] DEBUG[23743]: channel.c:3682 ast_generic_bridge: Didn't get a frame from channel: SIP/5060-08ad4f40 [Jan 15 15:35:15] DEBUG[23743]: channel.c:3986 ast_channel_bridge: Bridge stops bridging channels SIP/5060-08ad4f40 and SIP/216.226.69.244-08bdee68 [Jan 15 15:35:15] DEBUG[23743]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-08bdee68' [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-08bdee68, SIP callid 4938b4c60b8b46e4513941bb57c368a1@217.113 .64.11) [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:3278 sip_hangup: update_call_counter(2209847094) - decrement call limit counter on hangup [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call Scheduling destruction of SIP dialog '4938b4c60b8b46e4513941bb57c368a1@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:5576 reqprep: Strict routing enforced for session 4938b4c60b8b46e4513941bb57c368a1@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:2209847094@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5501af0e;rport From: "393200364746" ;tag=as627b2bfe To: ;tag=t1168871481-co1079 Call-ID: 4938b4c60b8b46e4513941bb57c368a1@217.113.64.11 CSeq: 103 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10956 [Jan 15 15:35:15] DEBUG[23743]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-08bdee68 [Jan 15 15:35:15] DEBUG[23743]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:15] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:15] DEBUG[23743]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 15 15:35:15] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:15] DEBUG[23743]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,0432209847094,1) exited non-zero on 'SIP/5060-08ad4f40' == Spawn extension (default, 0432209847094, 1) exited non-zero on 'SIP/5060-08ad4f40' [Jan 15 15:35:15] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:15] DEBUG[23743]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:15] DEBUG[24307]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:15] DEBUG[23743]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08ad4f40' [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08ad4f40, SIP callid 6994090221311200461-1168871480@172.16.100.100) [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:15] DEBUG[23743]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:15] DEBUG[23743]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08ad4f40 [Jan 15 15:35:15] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:15] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:15] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:15] DEBUG[24308]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. Really destroying SIP dialog '6994090221311200461-1168871480@172.16.100.100' Method: BYE [Jan 15 15:35:15] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '6994090221311200751-1168871675@172.16.100.100' [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 6994090221311200751-1168871675@172.16.100.100 Really destroying SIP dialog '6994090221311200751-1168871675@172.16.100.100' Method: ACK [Jan 15 15:35:15] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '4bfdf0694cef7bec2b3ccbf569eacbc8@217.113.64.11' [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 4bfdf0694cef7bec2b3ccbf569eacbc8@217.113.64.11 Really destroying SIP dialog '4bfdf0694cef7bec2b3ccbf569eacbc8@217.113.64.11' Method: INVITE [Jan 15 15:35:15] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[22864]: 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=z9hG4bK5501af0e To: ;tag=t1168871481-co1079 From: "393200364746" ;tag=as627b2bfe Call-ID: 4938b4c60b8b46e4513941bb57c368a1@217.113.64.11 CSeq: 103 BYE User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1079-CPO02208 Content-Length: 0 <-------------> [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5501af0e (64) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871481-co1079 (58) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "393200364746" ;tag=as627b2bfe (68) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 4938b4c60b8b46e4513941bb57c368a1@217.113.64.11 (55) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 103 BYE (13) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO1079-CPO02208 (61) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10956 [Jan 15 15:35:15] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '4938b4c60b8b46e4513941bb57c368a1@217.113.64.11' of Request 103: M atch Not Found Really destroying SIP dialog '4938b4c60b8b46e4513941bb57c368a1@217.113.64.11' Method: INVITE [Jan 15 15:35:15] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:15] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:15] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK45c475c6 To: ;tag=t1168871713-co170 From: "31643659444" ;tag=as1fa5c7b9 Call-ID: 18cc90fe7207502069e344106bf9f53e@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO170-CPO01942 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1636882176 1636882176 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 54036 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 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK45c475c6 (64) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871713-co170 (57) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "31643659444" ;tag=as1fa5c7b9 (66) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 18cc90fe7207502069e344106bf9f53e@217.113.64.11 (55) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO170-CPO01942 (60) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1636882176 1636882176 IN IP4 216.226.69.244 (47) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 193.19.106.196 (23) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 54036 RTP/AVP 18 101 (28) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '18cc90fe7207502069e3441 06bf9f53e@217.113.64.11' Request 102: Found [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 193.19.106.196:54036 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-091184f8 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 193.19.106.196:54036 [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-091184f8 is making progress passing it to SIP/5060-08bd4250 [Jan 15 15:35:16] DEBUG[24299]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:16] DEBUG[24299]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:16] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #10914 (4) SIP/2.0 - 1 [Jan 15 15:35:16] DEBUG[16996]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #10914)) 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@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 15 15:35:16] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[24293]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:16] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:16] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:17] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:17] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:17] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209961440@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200837-1168871717@172.16.100.100 From: ;tag=6116 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e45-ac106464-3455384 Contact: sip:442075366450@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 220 v=0 o=MG4000|2.0 874 874 IN IP4 10.57.3.21 s=- c=IN IP4 10.57.3.21 t=0 0 m=audio 21136 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 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209961440@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200837-1168871717@172.16.100.100 (54) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=6116 (64) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e45-ac106464-3 455384 (85) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:442075366450@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 220 (19) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 874 874 IN IP4 10.57.3.21 (38) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.21 (19) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 21136 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200837-1168871717@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:17] DEBUG[16996]: 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 - 6994090221311200837-1168871717@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:17] DEBUG[16996]: 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 15 15:35:17] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.21:21136 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 15 15:35:17] DEBUG[16996]: 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 10.57.3.21:21136 [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209961440 in default (domain 10.100.20.11) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:442075366450@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-0898d510: 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-6110000000498e45-ac106464-3455384;received=172.16.100.100 From: ;tag=6116 To: Call-ID: 6994090221311200837-1168871717@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 15 15:35:17] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0898d510 [Jan 15 15:35:17] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:17] DEBUG[24309]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432209961440@default:1] Dial("SIP/5060-0898d510", "SIP/2209961440@216.226.69.244||t") in new stack [Jan 15 15:35:17] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:17] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:17] DEBUG[24310]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:17] DEBUG[24309]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432209961440-1. [Jan 15 15:35:17] DEBUG[24309]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:17] DEBUG[24309]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:17] DEBUG[24309]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:2804 sip_call: Outgoing Call for 2209961440 [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 14408 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209961440@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK6bdcffe6;rport (64) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 2: From: "442075366450" ;tag=as6be5cd87 (68) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 3c23698b447c551b5c1bb6f30d6c42e7@217.113.64.11 (55) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:17 GMT (35) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: m=audio 14408 RTP/AVP 18 (24) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209961440@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK6bdcffe6;rport From: "442075366450" ;tag=as6be5cd87 To: Contact: Call-ID: 3c23698b447c551b5c1bb6f30d6c42e7@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 14408 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:17] DEBUG[24309]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10958 -- Called 2209961440@216.226.69.244 [Jan 15 15:35:17] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:17] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[24293]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 15 15:35:17] DEBUG[24161]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:17] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:17] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432207827776@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200839-1168871717@172.16.100.100 From: ;tag=6116 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e47-ac106464-3455385 Contact: sip:1214299642@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 1055 1055 IN IP4 10.57.3.25 s=- c=IN IP4 10.57.3.25 t=0 0 m=audio 25956 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 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432207827776@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200839-1168871717@172.16.100.100 (54) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=6116 (62) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e47-ac106464-3 455385 (85) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:1214299642@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 1055 1055 IN IP4 10.57.3.25 (40) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.25 (19) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 25956 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200839-1168871717@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:17] DEBUG[16996]: 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 - 6994090221311200839-1168871717@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:17] DEBUG[16996]: 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 15 15:35:17] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.25:25956 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 15 15:35:17] DEBUG[16996]: 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 10.57.3.25:25956 [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432207827776 in default (domain 10.100.20.11) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:1214299642@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:17] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-08726ac8: 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-6110000000498e47-ac106464-3455385;received=172.16.100.100 From: ;tag=6116 To: Call-ID: 6994090221311200839-1168871717@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 15 15:35:17] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08726ac8 [Jan 15 15:35:17] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:17] DEBUG[24311]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432207827776@default:1] Dial("SIP/5060-08726ac8", "SIP/2207827776@216.226.69.244||t") in new stack [Jan 15 15:35:17] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:17] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:17] DEBUG[24312]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:17] DEBUG[24311]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432207827776-1. [Jan 15 15:35:17] DEBUG[24311]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:17] DEBUG[24311]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:17] DEBUG[24311]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:2804 sip_call: Outgoing Call for 2207827776 [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 19300 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2207827776@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK03c9ccb7;rport (64) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 2: From: "1214299642" ;tag=as7417cd6c (64) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 18b10d2320a2a0e2574a92f45328789f@217.113.64.11 (55) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:17 GMT (35) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: m=audio 19300 RTP/AVP 18 (24) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2207827776@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK03c9ccb7;rport From: "1214299642" ;tag=as7417cd6c To: Contact: Call-ID: 18b10d2320a2a0e2574a92f45328789f@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 19300 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:17] DEBUG[24311]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10960 -- Called 2207827776@216.226.69.244 [Jan 15 15:35:17] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:17] DEBUG[23173]: 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=z9hG4bK5206503b To: ;tag=t1168871712-co160 From: "972641064" ;tag=as1883046c Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO160-CPO01918 Content-Type: application/sdp Content-Length: 228 v=0 o=- 438700668 438700668 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 41704 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 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5206503b (64) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871712-co160 (57) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972641064" ;tag=as1883046c (62) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 (55) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO160-CPO01918 (60) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 438700668 438700668 IN IP4 216.226.69.244 (45) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 41704 RTP/AVP 18 101 (28) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '796ca1b868f7cce6709aafe c493b6158@217.113.64.11' Request 102: Found [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:41704 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0868fbc8 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 216.226.69.237:41704 [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-0868fbc8 is making progress passing it to SIP/5060-0868e648 [Jan 15 15:35:18] DEBUG[24293]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:18] DEBUG[24293]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:18] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24138]: 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=z9hG4bK6bdcffe6 To: From: "442075366450" ;tag=as6be5cd87 Call-ID: 3c23698b447c551b5c1bb6f30d6c42e7@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO190-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK6bdcffe6 (64) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "442075366450" ;tag=as6be5cd87 (68) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 3c23698b447c551b5c1bb6f30d6c42e7@217.113.64.11 (55) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO190-CPO00000 (57) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10958 - INVITE (got response) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3c23698b447c551b5c1bb6f 30d6c42e7@217.113.64.11' Request 102: Found [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:18] DEBUG[24309]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:18] DEBUG[24309]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:18] DEBUG[24309]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:18] DEBUG[24309]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15956 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:18] DEBUG[24309]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:18] DEBUG[24309]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- 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-6110000000498e45-ac106464-3455384;received=172.16.100.100 From: ;tag=6116 To: ;tag=as2f490e5e Call-ID: 6994090221311200837-1168871717@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15956 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 15 15:35:18] DEBUG[24309]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:18] DEBUG[24309]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:18] DEBUG[24309]: rtp.c:870 ast_rtcp_read: Got RTCP report of 52 bytes [Jan 15 15:35:18] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24161]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24044]: 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=z9hG4bK03c9ccb7 To: From: "1214299642" ;tag=as7417cd6c Call-ID: 18b10d2320a2a0e2574a92f45328789f@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO199-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK03c9ccb7 (64) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "1214299642" ;tag=as7417cd6c (64) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 18b10d2320a2a0e2574a92f45328789f@217.113.64.11 (55) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO199-CPO00000 (57) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10960 - INVITE (got response) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '18b10d2320a2a0e2574a92f 45328789f@217.113.64.11' Request 102: Found [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:18] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[24311]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:18] DEBUG[24311]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:18] DEBUG[24311]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:18] DEBUG[24311]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15696 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:18] DEBUG[24311]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:18] DEBUG[24311]: 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-6110000000498e47-ac106464-3455385;received=172.16.100.100 From: ;tag=6116 To: ;tag=as7a7dc455 Call-ID: 6994090221311200839-1168871717@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15696 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 15 15:35:18] DEBUG[24311]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:18] DEBUG[24311]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:18] DEBUG[24309]: rtp.c:870 ast_rtcp_read: Got RTCP report of 48 bytes [Jan 15 15:35:18] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24297]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432205667373@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200841-1168871718@172.16.100.100 From: ;tag=22954 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e49-ac106464-3455386 Contact: sip:395459803620@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 7850 7850 IN IP4 10.57.3.22 s=- c=IN IP4 10.57.3.22 t=0 0 m=audio 60012 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 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432205667373@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200841-1168871718@172.16.100.100 (54) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=22954 (65) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e49-ac106464-3 455386 (85) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:395459803620@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 7850 7850 IN IP4 10.57.3.22 (40) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.22 (19) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 60012 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200841-1168871718@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:18] DEBUG[16996]: 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 - 6994090221311200841-1168871718@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:18] DEBUG[16996]: 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 15 15:35:18] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.22:60012 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 15 15:35:18] DEBUG[16996]: 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 10.57.3.22:60012 [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432205667373 in default (domain 10.100.20.11) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:395459803620@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-087eaf80: 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-6110000000498e49-ac106464-3455386;received=172.16.100.100 From: ;tag=22954 To: Call-ID: 6994090221311200841-1168871718@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 15 15:35:18] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-087eaf80 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:18] DEBUG[24313]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432205667373@default:1] Dial("SIP/5060-087eaf80", "SIP/2205667373@216.226.69.244||t") in new stack [Jan 15 15:35:18] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:18] DEBUG[24314]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:18] DEBUG[24313]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432205667373-1. [Jan 15 15:35:18] DEBUG[24313]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:18] DEBUG[24313]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:18] DEBUG[24313]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:2804 sip_call: Outgoing Call for 2205667373 [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 17904 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2205667373@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5a445744;rport (64) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 2: From: "395459803620" ;tag=as564d41ab (68) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 (55) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:18 GMT (35) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: m=audio 17904 RTP/AVP 18 (24) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2205667373@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5a445744;rport From: "395459803620" ;tag=as564d41ab To: Contact: Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 17904 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:18] DEBUG[24313]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10965 -- Called 2205667373@216.226.69.244 <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5362c6d6 To: ;tag=t1168871713-co169 From: "4921199999" ;tag=as452eaf89 Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 CSeq: 102 INVITE Reason: Q.850 ;cause=41 ;text="Unknown" Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO169-CPO01939 Content-Length: 0 <-------------> [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 503 Service Unavailable (31) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5362c6d6 (64) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871713-co169 (57) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as452eaf89 (64) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 (55) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850 ;cause=41 ;text="Unknown" (39) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: (45) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO169-CPO01939 (60) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 0 (17) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '0fe9f566534b27380d95344c3bfb1d95@217.113.64.11' of Request 102: M atch Not Found -- Got SIP response 503 "Service Unavailable" back from 216.226.69.244 Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2205671490@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5362c6d6;rport From: "4921199999" ;tag=as452eaf89 To: ;tag=t1168871713-co169 Contact: Call-ID: 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 0fe9f566534b27380d95344c3bfb1d95@217.113.64.11 -- SIP/216.226.69.244-08c7db18 is circuit-busy [Jan 15 15:35:18] DEBUG[24297]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-08c7db18' [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-08c7db18, SIP callid 0fe9f566534b27380d95344c3bfb1d95@217.113 .64.11) [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:3278 sip_hangup: update_call_counter(2205671490) - decrement call limit counter on hangup [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:18] DEBUG[24297]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-08c7db18 == Everyone is busy/congested at this time (1:0/1/0) [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:18] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:18] DEBUG[24297]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:18] DEBUG[24297]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. [Jan 15 15:35:18] DEBUG[24315]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:18] DEBUG[24297]: pbx.c:1767 pbx_extension_helper: Launching 'Goto' -- Executing [0432205671490@default:2] Goto("SIP/5060-08ea38a8", "s-CONGESTION|1") in new stack -- Goto (default,s-CONGESTION,1) [Jan 15 15:35:18] DEBUG[24297]: pbx.c:1767 pbx_extension_helper: Launching 'Hangup' -- Executing [s-CONGESTION@default:1] Hangup("SIP/5060-08ea38a8", "") in new stack [Jan 15 15:35:18] DEBUG[24297]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,s-CONGESTION,1) exited non-zero on 'SIP/5060-08ea38a8' == Spawn extension (default, s-CONGESTION, 1) exited non-zero on 'SIP/5060-08ea38a8' [Jan 15 15:35:18] DEBUG[24297]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:18] DEBUG[24297]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08ea38a8' [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08ea38a8, SIP callid 6994090221311200827-1168871713@172.16.100.100) [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:18] DEBUG[24297]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311200827-1168871713@172.16.100.100' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3b-ac106464-3455378;received=172.16.100.100 From: ;tag=4299 To: ;tag=as197d3bd7 Call-ID: 6994090221311200827-1168871713@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 15 15:35:18] DEBUG[24297]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10968 [Jan 15 15:35:18] DEBUG[24297]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08ea38a8 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:18] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:18] DEBUG[24316]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200827-1168871713@172.16.100.100 From: ;tag=4299 To: ;tag=as197d3bd7 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3b-ac106464-3455378 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205671490@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200827-1168871713@172.16.100.100 (54) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (62) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as197d3bd7 (67) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3b-ac106464-3 455378 (85) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10968 [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200827-1168871713@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '0fe9f566534b27380d95344c3bfb1d95@217.113.64.11' Method: INVITE [Jan 15 15:35:18] DEBUG[24299]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:18] DEBUG[24311]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[24293]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432207728414@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200825-1168871711@172.16.100.100 From: ;tag=3390 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e39-ac106464-3455377 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432207728414@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200825-1168871711@172.16.100.100 (54) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=3390 (61) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e39-ac106464-3 455377 (85) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311200825-1168871711@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-6110000000498e39-ac106464-3455377;received=172.16.100.100 From: ;tag=3390 To: ;tag=as191dc5fc Call-ID: 6994090221311200825-1168871711@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 15 15:35:18] DEBUG[16996]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10969 <--- 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-6110000000498e39-ac106464-3455377;received=172.16.100.100 From: ;tag=3390 To: ;tag=as191dc5fc Call-ID: 6994090221311200825-1168871711@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 15 15:35:18] DEBUG[24293]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:18] DEBUG[24293]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-0868fbc8' [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-0868fbc8, SIP callid 796ca1b868f7cce6709aafec493b6158@217.113 .64.11) [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:3278 sip_hangup: update_call_counter(2207728414) - decrement call limit counter on hangup [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '796ca1b868f7cce6709aafec493b6158@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '796ca1b868f7cce6709aafec493b6158@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2207728414@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5206503b;rport From: "972641064" ;tag=as1883046c To: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10971 Scheduling destruction of SIP dialog '796ca1b868f7cce6709aafec493b6158@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:18] DEBUG[24293]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0868fbc8 [Jan 15 15:35:18] DEBUG[24293]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:18] DEBUG[24293]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,0432207728414,1) exited non-zero on 'SIP/5060-0868e648' == Spawn extension (default, 0432207728414, 1) exited non-zero on 'SIP/5060-0868e648' [Jan 15 15:35:18] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:18] DEBUG[24293]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:18] DEBUG[24317]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:18] DEBUG[24293]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-0868e648' [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-0868e648, SIP callid 6994090221311200825-1168871711@172.16.100.100) [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:18] DEBUG[24293]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 15 15:35:18] DEBUG[24293]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0868e648 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:18] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:18] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:18] DEBUG[24318]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432207728414@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200825-1168871711@172.16.100.100 From: ;tag=3390 To: ;tag=as191dc5fc CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e39-ac106464-3455377 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432207728414@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200825-1168871711@172.16.100.100 (54) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=3390 (61) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as191dc5fc (67) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e39-ac106464-3 455377 (85) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10969 [Jan 15 15:35:18] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200825-1168871711@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311200825-1168871711@172.16.100.100' Method: ACK [Jan 15 15:35:18] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:18] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:19] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- 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=z9hG4bK6bdcffe6 To: ;tag=t1168871717-co190 From: "442075366450" ;tag=as6be5cd87 Call-ID: 3c23698b447c551b5c1bb6f30d6c42e7@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO190-CPO02008 Content-Type: application/sdp Content-Length: 228 v=0 o=- 154784045 154784045 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 193.19.106.196 t=0 0 m=audio 52052 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 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK6bdcffe6 (64) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871717-co190 (57) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "442075366450" ;tag=as6be5cd87 (68) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 3c23698b447c551b5c1bb6f30d6c42e7@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO190-CPO02008 (60) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:19] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:19] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[23636]: 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=z9hG4bK5a445744 To: From: "395459803620" ;tag=as564d41ab Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO200-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5a445744 (64) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "395459803620" ;tag=as564d41ab (68) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO200-CPO00000 (57) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10965 - INVITE (got response) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '658c624977ccfaab01016cd 73d24d663@217.113.64.11' Request 102: Found [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:19] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[24313]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:19] DEBUG[24313]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:19] DEBUG[24313]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:19] DEBUG[24313]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15718 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:19] DEBUG[24313]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:19] DEBUG[24313]: 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-6110000000498e49-ac106464-3455386;received=172.16.100.100 From: ;tag=22954 To: ;tag=as6354cf6c Call-ID: 6994090221311200841-1168871718@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15718 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 15 15:35:19] DEBUG[24313]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:19] DEBUG[24313]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:19] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[23913]: 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=z9hG4bK5206503b To: ;tag=t1168871712-co160 From: "972641064" ;tag=as1883046c Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO160-CPO01918 Content-Length: 0 <-------------> [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5206503b (64) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871712-co160 (57) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972641064" ;tag=as1883046c (62) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO160-CPO01918 (60) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10971 [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '796ca1b868f7cce6709aafec493b6158@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=z9hG4bK5206503b To: ;tag=t1168871712-co160 From: "972641064" ;tag=as1883046c Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO160-CPO01918 Content-Length: 0 <-------------> [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5206503b (64) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871712-co160 (57) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972641064" ;tag=as1883046c (62) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO160-CPO01918 (60) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '796ca1b868f7cce6709aafec493b6158@217.113.64.11' of Request 102: M atch Found [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2207728414@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK5206503b;rport From: "972641064" ;tag=as1883046c To: ;tag=t1168871712-co160 Contact: Call-ID: 796ca1b868f7cce6709aafec493b6158@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:19] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:19] DEBUG[24313]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:19] DEBUG[24303]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:02642491422524@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200843-1168871719@172.16.100.100 From: ;tag=7024 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e4b-ac106464-3455388 Contact: sip:2356476756@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 9467 9467 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 55356 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 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:02642491422524@10.100.20.11:5060;user=phone SIP/2.0 (62) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200843-1168871719@172.16.100.100 (54) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=7024 (62) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (53) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e4b-ac106464-3 455388 (85) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356476756@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9467 9467 IN IP4 10.57.2.121 (41) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 55356 RTP/AVP 18 101 13 (31) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200843-1168871719@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:19] DEBUG[16996]: 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 - 6994090221311200843-1168871719@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:19] DEBUG[16996]: 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 15 15:35:19] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:55356 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 15 15:35:19] DEBUG[16996]: 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 10.57.2.121:55356 [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 02642491422524 in default (domain 10.100.20.11) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:2356476756@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-0868e648: 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-6110000000498e4b-ac106464-3455388;received=172.16.100.100 From: ;tag=7024 To: Call-ID: 6994090221311200843-1168871719@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 15 15:35:19] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-0868e648 [Jan 15 15:35:19] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:19] DEBUG[24319]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [02642491422524@default:1] Dial("SIP/5060-0868e648", "SIP/2491422524@196.29.163.6||t") in new stack [Jan 15 15:35:19] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:19] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:19] DEBUG[24320]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:19] DEBUG[24319]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-02642491422524-1. [Jan 15 15:35:19] DEBUG[24319]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:19] DEBUG[24319]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:19] DEBUG[24319]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:2804 sip_call: Outgoing Call for 2491422524 [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 10846 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2491422524@196.29.163.6 SIP/2.0 (42) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK24e12220;rport (64) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 2: From: "2356476756" ;tag=as001c6a5c (64) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 3: To: (33) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 4: Contact: (39) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 1066854d67b118413ad990e04882847e@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:19 GMT (35) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: m=audio 10846 RTP/AVP 18 (24) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:2491422524@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK24e12220;rport From: "2356476756" ;tag=as001c6a5c To: Contact: Call-ID: 1066854d67b118413ad990e04882847e@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 10846 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:19] DEBUG[24319]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10975 -- Called 2491422524@196.29.163.6 [Jan 15 15:35:19] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:19] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:19] DEBUG[24216]: 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: "2356476756" ;tag=as001c6a5c To: CSeq: 102 INVITE Call-ID: 1066854d67b118413ad990e04882847e@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK24e12220;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "2356476756" ;tag=as001c6a5c (64) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (33) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 1066854d67b118413ad990e04882847e@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK24e12220;rport=5060 (69) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10975 - INVITE (got response) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1066854d67b118413ad990e 04882847e@217.113.64.11' Request 102: Found [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:19] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5a445744 To: ;tag=t1168871719-co200 From: "395459803620" ;tag=as564d41ab Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO200-CPO02051 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1694188404 1694188404 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40248 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 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5a445744 (64) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871719-co200 (57) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "395459803620" ;tag=as564d41ab (68) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 (55) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO200-CPO02051 (60) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1694188404 1694188404 IN IP4 216.226.69.244 (47) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40248 RTP/AVP 18 101 (28) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '658c624977ccfaab01016cd 73d24d663@217.113.64.11' Request 102: Found [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40248 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0894c918 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 216.226.69.237:40248 [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:19] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-0894c918 is making progress passing it to SIP/5060-087eaf80 [Jan 15 15:35:19] DEBUG[24313]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:19] DEBUG[24313]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:19] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:20] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 180 Ringing From: "2356476756" ;tag=as001c6a5c To: ;tag=84dfb805 CSeq: 102 INVITE Call-ID: 1066854d67b118413ad990e04882847e@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK24e12220;rport=5060 Contact: Content-Length: 151 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 10560246 10560246 IN IP4 10.10.13.4 s=Sip Call c=IN IP4 196.29.163.6 t=0 0 m=audio 40628 RTP/AVP 18 a=rtpmap:18 G729/8000 <-------------> [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "2356476756" ;tag=as001c6a5c (64) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=84dfb805 (46) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 1066854d67b118413ad990e04882847e@217.113.64.11 (55) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK24e12220;rport=5060 (69) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (54) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 151 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=HuaweiSoftX3000 10560246 10560246 IN IP4 10.10.13.4 (53) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=Sip Call (10) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 196.29.163.6 (21) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40628 RTP/AVP 18 (24) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) --- (9 headers 7 lines) --- [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1066854d67b118413ad990e 04882847e@217.113.64.11' Request 102: Found [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 180 to standard invite [Jan 15 15:35:20] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-08bdee68 Found RTP audio format 18 [Jan 15 15:35:20] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 15 15:35:20] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 196.29.163.6:40628 [Jan 15 15:35:20] DEBUG[16977]: channel.c:943 channel_find_locked: Avoiding initial deadlock for channel '0x8e207b0' Found description format G729 for ID 18 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/196.29.163.6-08bdee68 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 196.29.163.6:40628 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/196.29.163.6-08bdee68 is ringing [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #10914 (5) SIP/2.0 - 1 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #10914)) 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@172.16.100.100 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- 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-6110000000498e4b-ac106464-3455388;received=172.16.100.100 From: ;tag=7024 To: ;tag=as288faa06 Call-ID: 6994090221311200843-1168871719@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/196.29.163.6-08bdee68 is making progress passing it to SIP/5060-0868e648 [Jan 15 15:35:20] DEBUG[24319]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:20] DEBUG[24319]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:20] DEBUG[24319]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 19582 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:20] DEBUG[24319]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:20] DEBUG[24319]: 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-6110000000498e4b-ac106464-3455388;received=172.16.100.100 From: ;tag=7024 To: ;tag=as288faa06 Call-ID: 6994090221311200843-1168871719@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 19582 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 15 15:35:20] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:20] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 6 (Ringing) [Jan 15 15:35:20] DEBUG[24321]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jan 15 15:35:20] DEBUG[24319]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:20] DEBUG[24319]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:20] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:20] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[24303]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes <--- SIP read from 216.226.69.244:5060 ---> INVITE sip:218214630913@217.113.64.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 216.226.69.244:5060;branch=z9hG4bK1d8e245f445ab90f4-6d6-1 To: "218214630913" ;tag=as3596131c From: ;tag=t1168871668-co1750 Date: Mon, 15 Jan 2007 14:35:20 GMT Call-ID: 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 CSeq: 77610 INVITE Max-Forwards: 70 Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1750-CPO00894 Content-Type: application/sdp Content-Length: 375 v=0 o=- 898387760 898387761 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=image 44074 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxMaxBuffer:524 a=T38FaxMaxDatagram:176 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv <-------------> [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:218214630913@217.113.64.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 216.226.69.244:5060;branch=z9hG4bK1d8e245f445ab90f4-6d6-1 (74) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: "218214630913" ;tag=as3596131c (66) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: ;tag=t1168871668-co1750 (60) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Date: Mon, 15 Jan 2007 14:35:20 GMT (36) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 (55) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: CSeq: 77610 INVITE (18) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Contact: (47) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1750-CPO00894 (61) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Content-Type: application/sdp (29) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 375 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 898387760 898387761 IN IP4 216.226.69.244 (45) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=image 44074 udptl t38 (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxVersion:0 (17) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38MaxBitRate:9600 (20) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxMaxBuffer:524 (21) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxMaxDatagram:176 (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (12 headers 16 lines) --- [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 216.226.69.244 : 5060 (no NAT) Got T.38 offer in SDP in dialog 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4778 process_sdp: T38 state changed to 4 on channel SIP/216.226.69.244-08c27e20 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4828 process_sdp: Peer T.38 UDPTL is at port 216.226.69.237:44074 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4994 process_sdp: FaxVersion: 0 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4970 process_sdp: T38MaxBitRate: 9600 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4966 process_sdp: MaxBufferSize:524 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5002 process_sdp: FaxMaxDatagram: 176 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5027 process_sdp: RateMangement: transferredTCF [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5008 process_sdp: FillBitRemoval: 0 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5014 process_sdp: Transcoding MMR: 0 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5021 process_sdp: Transcoding JBIG: 0 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5035 process_sdp: UDP EC: t38UDPRedundancy [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5055 process_sdp: Our T38 capability = (3872), peer T38 capability (3872), joint T38 capability (3872) Capabilities: us - 0x100 (g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5097 process_sdp: Have T.38 but no audio codecs, accepting offer anyway [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:13343 handle_request_invite: Got a SIP re-invite for call 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:13438 handle_request_invite: SIP/216.226.69.244-08c27e20: This call is UP.... [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:16591 sip_handle_t38_reinvite: Sending reinvite on SIP '6994090221311200735-1168871667@172.16.100.100' - It's UDPTL soon redirected to us (IP 10.100.20.11) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 6994090221311200735-1168871667@172.16.100.100 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.100.100, port 5060 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5975 add_t38_sdp: T.38 UDPTL is at 10.100.20.11 port 18809 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5981 add_t38_sdp: Our T38 capability (3872), peer T38 capability (3872), joint capability (3872) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5908 t38_get_rate: T38MaxFaxRate 9600 found [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1597 initialize_initreq: Initializing already initialized SIP dialog 6994090221311200735-1168871667@172.16.10 0.100 (presumably reinvite) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:218214630913@172.16.100.100:5060;user=phone SIP/2.0 (62) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK269cd277;rport (63) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=as2237b570 (69) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=16168 (63) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 6994090221311200735-1168871667@172.16.100.100 (54) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Supported: replaces (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: X-asterisk-info: SIP re-invite (T38 switchover) (47) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 350 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=root 16974 16976 IN IP4 10.100.20.11 (38) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.100.20.11 (21) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=image 18809 udptl t38 (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxVersion:0 (17) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38MaxBitRate:9600 (20) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxMaxBuffer:176 (21) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxMaxDatagram:176 (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) Reliably Transmitting (no NAT) to 172.16.100.100:5060: INVITE sip:218214630913@172.16.100.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK269cd277;rport From: ;tag=as2237b570 To: ;tag=16168 Contact: Call-ID: 6994090221311200735-1168871667@172.16.100.100 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 350 v=0 o=root 16974 16976 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=image 18809 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy --- [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10980 [Jan 15 15:35:20] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes <--- SIP read from 172.16.100.100:5060 ---> SIP/2.0 200 OK Call-ID: 6994090221311200735-1168871667@172.16.100.100 From: ;tag=as2237b570 To: ;tag=16168 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK269cd277;rport Contact: sip:218214630913@172.16.100.100:5060;user=phone Supported: timer,100rel Content-Length: 105 v=0 o=MG4000|2.0 3639 3639 IN IP4 10.57.3.21 s=- c=IN IP4 10.57.3.21 t=0 0 m=image 29836 udptl t38 <-------------> [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200735-1168871667@172.16.100.100 (54) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=as2237b570 (69) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=16168 (63) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK269cd277;rport (63) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:218214630913@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Supported: timer,100rel (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 105 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 3639 3639 IN IP4 10.57.3.21 (40) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.21 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=image 29836 udptl t38 (23) --- (10 headers 6 lines) --- [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10980 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200735-1168871667@172.16.100.100' of Request 102: Ma tch Not Found [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:11537 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6994090221311200735-1168871667@1 72.16.100.100 Got T.38 offer in SDP in dialog 6994090221311200735-1168871667@172.16.100.100 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4778 process_sdp: T38 state changed to 4 on channel SIP/5060-08be3f28 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4828 process_sdp: Peer T.38 UDPTL is at port 10.57.3.21:29836 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 6994090221311200735-1168871667@172.16.100.100 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5055 process_sdp: Our T38 capability = (3872), peer T38 capability (3872), joint T38 capability (3872) Capabilities: us - 0x100 (g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5097 process_sdp: Have T.38 but no audio codecs, accepting offer anyway [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:218214630913@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:16619 sip_handle_t38_reinvite: Responding 200 OK on SIP '37c33e857879ffe043e4a5282baabd9c@217.113.64.11' - It 's UDPTL soon redirected to us (IP 217.113.64.11) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:16624 sip_handle_t38_reinvite: T38 changed state to 5 on channel SIP/5060-08be3f28 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:16625 sip_handle_t38_reinvite: T38 changed state to 5 on channel SIP/216.226.69.244-08c27e20 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5975 add_t38_sdp: T.38 UDPTL is at 217.113.64.11 port 9819 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5981 add_t38_sdp: Our T38 capability (3872), peer T38 capability (3872), joint capability (3872) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5908 t38_get_rate: T38MaxFaxRate 9600 found <--- Reliably Transmitting (no NAT) to 216.226.69.244:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.226.69.244:5060;branch=z9hG4bK1d8e245f445ab90f4-6d6-1;received=216.226.69.244 From: ;tag=t1168871668-co1750 To: "218214630913" ;tag=as3596131c Call-ID: 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 CSeq: 77610 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 16974 16975 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=image 9819 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy <------------> [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10981 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 6994090221311200735-1168871667@172.16.100.100 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.100.100, port 5060 Transmitting (no NAT) to 172.16.100.100:5060: ACK sip:218214630913@172.16.100.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.100.20.11:5060;branch=z9hG4bK2149f194;rport From: ;tag=as2237b570 To: ;tag=16168 Contact: Call-ID: 6994090221311200735-1168871667@172.16.100.100 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:20] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '6994090221311200779-1168871687@172.16.100.100' [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 6994090221311200779-1168871687@172.16.100.100 Really destroying SIP dialog '6994090221311200779-1168871687@172.16.100.100' Method: ACK [Jan 15 15:35:20] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:20] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249912852296@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200845-1168871720@172.16.100.100 From: ;tag=23863 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e4d-ac106464-3455390 Contact: sip:249912301506@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 220 v=0 o=MG4000|2.0 4207 4207 IN IP4 10.57.2.54 s=- c=IN IP4 10.57.2.54 t=0 0 m=audio 40604 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 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249912852296@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200845-1168871720@172.16.100.100 (54) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=23863 (65) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e4d-ac106464-3 455390 (85) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:249912301506@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 220 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 4207 4207 IN IP4 10.57.2.54 (40) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.54 (19) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40604 RTP/AVP 18 101 13 (31) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200845-1168871720@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:20] DEBUG[16996]: 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 - 6994090221311200845-1168871720@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:20] DEBUG[16996]: 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 15 15:35:20] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.54:40604 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 15 15:35:20] DEBUG[16996]: 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 10.57.2.54:40604 [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249912852296 in default (domain 10.100.20.11) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:249912301506@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:20] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-084d9978: 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-6110000000498e4d-ac106464-3455390;received=172.16.100.100 From: ;tag=23863 To: Call-ID: 6994090221311200845-1168871720@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 15 15:35:20] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-084d9978 [Jan 15 15:35:20] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:20] DEBUG[24322]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0264249912852296@default:1] Dial("SIP/5060-084d9978", "SIP/249912852296@196.29.163.6||t") in new stack [Jan 15 15:35:20] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:20] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:20] DEBUG[24323]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:20] DEBUG[24322]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0264249912852296-1. [Jan 15 15:35:20] DEBUG[24322]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:20] DEBUG[24322]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:20] DEBUG[24322]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:2804 sip_call: Outgoing Call for 249912852296 [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 11064 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249912852296@196.29.163.6 SIP/2.0 (44) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0a610aee;rport (64) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 2: From: "249912301506" ;tag=as19fa0939 (68) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 08f411773d78de697c7d1ac41c2dc14a@217.113.64.11 (55) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:20 GMT (35) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: m=audio 11064 RTP/AVP 18 (24) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249912852296@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0a610aee;rport From: "249912301506" ;tag=as19fa0939 To: Contact: Call-ID: 08f411773d78de697c7d1ac41c2dc14a@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 11064 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:20] DEBUG[24322]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10982 -- Called 249912852296@196.29.163.6 [Jan 15 15:35:20] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[24319]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:21] DEBUG[24319]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:21] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[24319]: 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: "249912301506" ;tag=as19fa0939 To: CSeq: 102 INVITE Call-ID: 08f411773d78de697c7d1ac41c2dc14a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0a610aee;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "249912301506" ;tag=as19fa0939 (68) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 08f411773d78de697c7d1ac41c2dc14a@217.113.64.11 (55) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0a610aee;rport=5060 (69) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #10982 - INVITE (got response) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '08f411773d78de697c7d1ac 41c2dc14a@217.113.64.11' Request 102: Found [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:21] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes <--- SIP read from 216.226.69.244:5060 ---> ACK sip:218214630913@217.113.64.11:5060 SIP/2.0 Via: SIP/2.0/UDP 216.226.69.244:5060;branch=z9hG4bK1d8e245f445ab90f4-6d6-2 To: "218214630913" ;tag=as3596131c From: ;tag=t1168871668-co1750 Call-ID: 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 CSeq: 77610 ACK Max-Forwards: 70 User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1750-CPO00894 Content-Length: 0 <-------------> [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:218214630913@217.113.64.11:5060 SIP/2.0 (47) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 216.226.69.244:5060;branch=z9hG4bK1d8e245f445ab90f4-6d6-2 (74) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: "218214630913" ;tag=as3596131c (66) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: ;tag=t1168871668-co1750 (60) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 37c33e857879ffe043e4a5282baabd9c@217.113.64.11 (55) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 77610 ACK (15) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO1750-CPO00894 (61) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10981 [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '37c33e857879ffe043e4a5282baabd9c@217.113.64.11' of Response 77610 : Match Not Found [Jan 15 15:35:21] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24299]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:21] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '6994090221311200749-1168871673@172.16.100.100' [Jan 15 15:35:21] DEBUG[16996]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 6994090221311200749-1168871673@172.16.100.100 Really destroying SIP dialog '6994090221311200749-1168871673@172.16.100.100' Method: ACK [Jan 15 15:35:22] DEBUG[24311]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 200 OK From: "24107371228" ;tag=as3c441b01 To: ;tag=26681538 CSeq: 102 INVITE Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK79f70944;rport=5060 Contact: Content-Length: 151 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 10560228 10560230 IN IP4 10.10.13.4 s=Sip Call c=IN IP4 196.29.163.6 t=0 0 m=audio 40626 RTP/AVP 18 a=rtpmap:18 G729/8000 <-------------> [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "24107371228" ;tag=as3c441b01 (66) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=26681538 (48) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 (55) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK79f70944;rport=5060 (69) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (56) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 151 (19) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=HuaweiSoftX3000 10560228 10560230 IN IP4 10.10.13.4 (53) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=Sip Call (10) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 196.29.163.6 (21) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40626 RTP/AVP 18 (24) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) --- (9 headers 7 lines) --- [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '22580e741be995b944ab373a6a35e55a@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 196.29.163.6:40626 Found description format G729 for ID 18 [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/196.29.163.6-08a029c8 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 196.29.163.6:40626 [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 22580e741be995b944ab373a6a35e55a@217.113.64.11 set_destination: Parsing for address/port to send to set_destination: set destination to 196.29.163.6, port 5060 Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249912495880@196.29.163.6:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK2058684f;rport From: "24107371228" ;tag=as3c441b01 To: ;tag=26681538 Contact: Call-ID: 22580e741be995b944ab373a6a35e55a@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:22] DEBUG[24263]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-08a029c8 -- SIP/196.29.163.6-08a029c8 answered SIP/5060-08586e70 [Jan 15 15:35:22] DEBUG[24263]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08586e70 [Jan 15 15:35:22] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 15 15:35:22] DEBUG[24263]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-08586e70 [Jan 15 15:35:22] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:22] DEBUG[24263]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:22] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) [Jan 15 15:35:22] DEBUG[24263]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:22] DEBUG[24324]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jan 15 15:35:22] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:22] DEBUG[24263]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11890 [Jan 15 15:35:22] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:22] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:22] DEBUG[24325]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:22] DEBUG[24263]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:22] DEBUG[24263]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) localhost*CLI> <--- 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-6110000000498e1f-ac106464-3455365;received=172.16.100.100 From: ;tag=32523 To: ;tag=as65d195fd Call-ID: 6994090221311200799-1168871703@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 16974 16975 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11890 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 15 15:35:22] DEBUG[24263]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10984 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0264249912495880@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200799-1168871703@172.16.100.100 From: ;tag=32523 To: ;tag=as65d195fd CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e1f-ac106464-3455365 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0264249912495880@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200799-1168871703@172.16.100.100 (54) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=32523 (64) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as65d195fd (70) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e1f-ac106464-3 455365 (85) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10984 [Jan 15 15:35:22] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200799-1168871703@172.16.100.100' of Response 1: Mat ch Not Found [Jan 15 15:35:22] DEBUG[24263]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:22] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:22] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:22] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:23] DEBUG[24309]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:23] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 216.226.69.244:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK03c9ccb7 To: ;tag=t1168871718-co199 From: "1214299642" ;tag=as7417cd6c Call-ID: 18b10d2320a2a0e2574a92f45328789f@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO199-CPO02036 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1355314626 1355314626 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40212 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 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK03c9ccb7 (64) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871718-co199 (57) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "1214299642" ;tag=as7417cd6c (64) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 18b10d2320a2a0e2574a92f45328789f@217.113.64.11 (55) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1002-RG575-EP0-CO199-CPO02036 (60) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1355314626 1355314626 IN IP4 216.226.69.244 (47) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40212 RTP/AVP 18 101 (28) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '18b10d2320a2a0e2574a92f 45328789f@217.113.64.11' Request 102: Found [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40212 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08a469e8 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 216.226.69.237:40212 [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/216.226.69.244-08a469e8 is making progress passing it to SIP/5060-08726ac8 [Jan 15 15:35:23] DEBUG[24311]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:23] DEBUG[24311]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:23] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:23] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:23] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24263]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 15 15:35:23] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:23] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24311]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 15 15:35:23] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[24299]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:23] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:1986 __sip_autodestruct: Auto destroying SIP dialog '7066fc3857da3bf6106e3f8848f2d23d@217.113.64.11' [Jan 15 15:35:23] DEBUG[16996]: chan_sip.c:3072 sip_destroy: Destroying SIP dialog 7066fc3857da3bf6106e3f8848f2d23d@217.113.64.11 Really destroying SIP dialog '7066fc3857da3bf6106e3f8848f2d23d@217.113.64.11' Method: INVITE [Jan 15 15:35:24] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[24163]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[23939]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[24263]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[16996]: chan_sip.c:1842 retrans_pkt: SIP TIMER: Rescheduling retransmission #10914 (6) SIP/2.0 - 1 [Jan 15 15:35:24] DEBUG[16996]: chan_sip.c:1856 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #10914)) 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-6110000000498e23-ac106464-3455367;received=172.16.100.100 From: ;tag=664 To: ;tag=as00423b80 Call-ID: 6994090221311200803-1168871705@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 15 15:35:24] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[24313]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[23913]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[24322]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:24] DEBUG[24322]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:24] DEBUG[24322]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 13048 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:24] DEBUG[24322]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:24] DEBUG[24322]: 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-6110000000498e4d-ac106464-3455390;received=172.16.100.100 From: ;tag=23863 To: ;tag=as3a9bfb4a Call-ID: 6994090221311200845-1168871720@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 13048 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 15 15:35:24] DEBUG[24322]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:24] DEBUG[24322]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:24] DEBUG[24303]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[24216]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:24] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:24] DEBUG[24319]: 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=z9hG4bK7e1fafe7 To: ;tag=t1168871710-co143 From: "972320621" ;tag=as506570ea Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO143-CPO01873 Content-Type: application/sdp Content-Length: 228 v=0 o=- 750509029 750509029 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40132 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 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7e1fafe7 (64) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871710-co143 (57) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972320621" ;tag=as506570ea (62) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 (55) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO143-CPO01873 (60) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 750509029 750509029 IN IP4 216.226.69.244 (45) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40132 RTP/AVP 18 101 (28) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40132 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-08801420 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 216.226.69.237:40132 [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 62ed0c706f8b2b2d07a2e30658a0b16b@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:2208901690@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7430fc24;rport From: "972320621" ;tag=as506570ea To: ;tag=t1168871710-co143 Contact: Call-ID: 62ed0c706f8b2b2d07a2e30658a0b16b@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:25] DEBUG[24288]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-08801420 -- SIP/216.226.69.244-08801420 answered SIP/5060-08e5cf60 [Jan 15 15:35:25] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:25] DEBUG[24288]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08e5cf60 [Jan 15 15:35:25] DEBUG[24288]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-08e5cf60 [Jan 15 15:35:25] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:25] DEBUG[24288]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:25] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:25] DEBUG[24288]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:25] DEBUG[24288]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15888 [Jan 15 15:35:25] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:25] DEBUG[24326]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. Adding codec 0x100 (g729) to SDP [Jan 15 15:35:25] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:25] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:25] DEBUG[24288]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:25] DEBUG[24327]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:25] DEBUG[24288]: 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-6110000000498e35-ac106464-3455375;received=172.16.100.100 From: ;tag=19320 To: ;tag=as614a422b Call-ID: 6994090221311200821-1168871710@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 16974 16975 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15888 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 15 15:35:25] DEBUG[24288]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10987 [Jan 15 15:35:25] DEBUG[24288]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432208901690@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200821-1168871710@172.16.100.100 From: ;tag=19320 To: ;tag=as614a422b CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e35-ac106464-3455375 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432208901690@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200821-1168871710@172.16.100.100 (54) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=19320 (62) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as614a422b (67) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e35-ac106464-3 455375 (85) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10987 [Jan 15 15:35:25] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200821-1168871710@172.16.100.100' of Response 1: Mat ch Not Found [Jan 15 15:35:25] DEBUG[24322]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[23943]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[20020]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[22394]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:25] DEBUG[24322]: rtp.c:1169 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Jan 15 15:35:25] DEBUG[24299]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:25] DEBUG[24322]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24224]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[23124]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249911243881@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200851-1168871725@172.16.100.100 From: ;tag=9750 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e53-ac106464-3455392 Contact: sip:256752538640@172.16.100.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 220 v=0 o=MG4000|2.0 4193 4193 IN IP4 10.57.2.54 s=- c=IN IP4 10.57.2.54 t=0 0 m=audio 20928 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 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249911243881@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200851-1168871725@172.16.100.100 (54) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=9750 (64) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e53-ac106464-3 455392 (85) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:256752538640@172.16.100.100:5060;user=phone (56) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 220 (19) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 4193 4193 IN IP4 10.57.2.54 (40) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.54 (19) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 20928 RTP/AVP 18 101 13 (31) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200851-1168871725@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:26] DEBUG[16996]: 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 - 6994090221311200851-1168871725@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:26] DEBUG[16996]: 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 15 15:35:26] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.54:20928 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 15 15:35:26] DEBUG[16996]: 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 10.57.2.54:20928 [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249911243881 in default (domain 10.100.20.11) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:256752538640@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-088bda18: 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-6110000000498e53-ac106464-3455392;received=172.16.100.100 From: ;tag=9750 To: Call-ID: 6994090221311200851-1168871725@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 15 15:35:26] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-088bda18 [Jan 15 15:35:26] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:26] DEBUG[24328]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0264249911243881@default:1] Dial("SIP/5060-088bda18", "SIP/249911243881@196.29.163.6||t") in new stack [Jan 15 15:35:26] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:26] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:26] DEBUG[24329]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:26] DEBUG[24328]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0264249911243881-1. [Jan 15 15:35:26] DEBUG[24328]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:26] DEBUG[24328]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:26] DEBUG[24328]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:2804 sip_call: Outgoing Call for 249911243881 [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 17142 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:249911243881@196.29.163.6 SIP/2.0 (44) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK44714c62;rport (64) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 2: From: "256752538640" ;tag=as290ce0f3 (68) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 4: Contact: (41) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 667a36a54544847e60a3666819395757@217.113.64.11 (55) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:26 GMT (35) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: m=audio 17142 RTP/AVP 18 (24) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 196.29.163.6:5060: INVITE sip:249911243881@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK44714c62;rport From: "256752538640" ;tag=as290ce0f3 To: Contact: Call-ID: 667a36a54544847e60a3666819395757@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 17142 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:26] DEBUG[24328]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10988 -- Called 249911243881@196.29.163.6 [Jan 15 15:35:26] DEBUG[23986]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24309]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24322]: rtp.c:870 ast_rtcp_read: Got RTCP report of 60 bytes [Jan 15 15:35:26] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[23422]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24311]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24002]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24138]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24303]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[22972]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24062]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[19799]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:26] DEBUG[24171]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> INFO sip:0432209709661@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311199525-1168870943@172.16.100.100 From: ;tag=14931 To: ;tag=as12e8d479 CSeq: 2 INFO Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498925-ac106464-3455393 Supported: timer,100rel Max-Forwards: 70 Content-Length: 11 Content-Type: application/dtmf-relay Signal= # <-------------> [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INFO sip:0432209709661@10.100.20.11:5060;user=phone SIP/2.0 (59) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311199525-1168870943@172.16.100.100 (54) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=14931 (62) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as12e8d479 (67) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 2 INFO (12) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498925-ac106464-3 455393 (85) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 11 (18) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Type: application/dtmf-relay (36) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: Signal= # (9) --- (10 headers 1 lines) --- [Jan 15 15:35:26] DEBUG[16996]: 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-6110000000498925-ac106464-3455393;received=172.16.100.100 From: ;tag=14931 To: ;tag=as12e8d479 Call-ID: 6994090221311199525-1168870943@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 15 15:35:26] DEBUG[23483]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24301]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes [Jan 15 15:35:26] DEBUG[23636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> CANCEL sip:0432204495604@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200831-1168871713@172.16.100.100 From: ;tag=4299 To: CSeq: 1 CANCEL Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3f-ac106464-3455380 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: CANCEL sip:0432204495604@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200831-1168871713@172.16.100.100 (54) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (62) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 CANCEL (14) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3f-ac106464-3 455380 (85) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received CANCEL (14) - Command in SIP CANCEL Sending to 172.16.100.100 : 5060 (no NAT) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311200831-1168871713@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-6110000000498e3f-ac106464-3455380;received=172.16.100.100 From: ;tag=4299 To: ;tag=as7c2ffa47 Call-ID: 6994090221311200831-1168871713@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 15 15:35:26] DEBUG[16996]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10990 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-6110000000498e3f-ac106464-3455380;received=172.16.100.100 From: ;tag=4299 To: ;tag=as7c2ffa47 Call-ID: 6994090221311200831-1168871713@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 15 15:35:26] DEBUG[24301]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:26] DEBUG[24301]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-08b01228' [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-08b01228, SIP callid 63d06de74913b81e3da871cc28c5a258@217.113 .64.11) [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:3278 sip_hangup: update_call_counter(2204495604) - decrement call limit counter on hangup [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Down (not UP) Scheduling destruction of SIP dialog '63d06de74913b81e3da871cc28c5a258@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '63d06de74913b81e3da871cc28c5a258@217.113.64.11' of Request 102: M atch Not Found Reliably Transmitting (no NAT) to 216.226.69.244:5060: CANCEL sip:2204495604@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7de4aefb;rport From: "4921199999" ;tag=as0028df42 To: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 CANCEL User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10992 Scheduling destruction of SIP dialog '63d06de74913b81e3da871cc28c5a258@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:26] DEBUG[24301]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-08b01228 [Jan 15 15:35:26] DEBUG[24301]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=CANCEL. [Jan 15 15:35:26] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:26] DEBUG[24301]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,0432204495604,1) exited non-zero on 'SIP/5060-08afd2c0' == Spawn extension (default, 0432204495604, 1) exited non-zero on 'SIP/5060-08afd2c0' [Jan 15 15:35:26] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:26] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:26] DEBUG[24301]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:26] DEBUG[24330]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:26] DEBUG[24301]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08afd2c0' [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08afd2c0, SIP callid 6994090221311200831-1168871713@172.16.100.100) [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:26] DEBUG[24301]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) [Jan 15 15:35:26] DEBUG[24301]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08afd2c0 [Jan 15 15:35:26] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:26] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:26] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:26] DEBUG[24331]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432204495604@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200831-1168871713@172.16.100.100 From: ;tag=4299 To: ;tag=as7c2ffa47 CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3f-ac106464-3455380 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432204495604@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200831-1168871713@172.16.100.100 (54) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=4299 (62) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as7c2ffa47 (67) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e3f-ac106464-3 455380 (85) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10990 [Jan 15 15:35:26] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200831-1168871713@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '6994090221311200831-1168871713@172.16.100.100' Method: ACK [Jan 15 15:35:26] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:26] DEBUG[24313]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 486 Busy Here From: "256752538640" ;tag=as290ce0f3 To: ;tag=7f01925d CSeq: 102 INVITE Call-ID: 667a36a54544847e60a3666819395757@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK44714c62;rport=5060 Reason: Q.850;cause=17;text="user busy" Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 486 Busy Here (21) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "256752538640" ;tag=as290ce0f3 (68) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=7f01925d (48) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 667a36a54544847e60a3666819395757@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK44714c62;rport=5060 (69) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Reason: Q.850;cause=17;text="user busy" (39) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10988 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '667a36a54544847e60a3666819395757@217.113.64.11' of Request 102: M atch Not Found -- Got SIP response 486 "Busy Here" back from 196.29.163.6 Transmitting (no NAT) to 196.29.163.6:5060: ACK sip:249911243881@196.29.163.6 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK44714c62;rport From: "256752538640" ;tag=as290ce0f3 To: ;tag=7f01925d Contact: Call-ID: 667a36a54544847e60a3666819395757@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 667a36a54544847e60a3666819395757@217.113.64.11 -- SIP/196.29.163.6-085bef78 is busy [Jan 15 15:35:27] DEBUG[24328]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/196.29.163.6-085bef78' [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:3270 sip_hangup: Hangup call SIP/196.29.163.6-085bef78, SIP callid 667a36a54544847e60a3666819395757@217.113.6 4.11) [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:3278 sip_hangup: update_call_counter(249911243881) - decrement call limit counter on hangup [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:27] DEBUG[24328]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-085bef78 == Everyone is busy/congested at this time (1:1/0/0) [Jan 15 15:35:27] DEBUG[24328]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:27] DEBUG[24328]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=BUSY. [Jan 15 15:35:27] DEBUG[24328]: pbx.c:1767 pbx_extension_helper: Launching 'Goto' -- Executing [0264249911243881@default:2] Goto("SIP/5060-088bda18", "s-BUSY|1") in new stack [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 -- Goto (default,s-BUSY,1) [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 2 (In use) [Jan 15 15:35:27] DEBUG[24328]: pbx.c:1767 pbx_extension_helper: Launching 'Hangup' -- Executing [s-BUSY@default:1] Hangup("SIP/5060-088bda18", "") in new stack [Jan 15 15:35:27] DEBUG[24332]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jan 15 15:35:27] DEBUG[24328]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,s-BUSY,1) exited non-zero on 'SIP/5060-088bda18' == Spawn extension (default, s-BUSY, 1) exited non-zero on 'SIP/5060-088bda18' [Jan 15 15:35:27] DEBUG[24328]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:27] DEBUG[24328]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-088bda18' [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-088bda18, SIP callid 6994090221311200851-1168871725@172.16.100.100) [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:27] DEBUG[24328]: chan_sip.c:3291 sip_hangup: Hanging up channel in state Ring (not UP) Scheduling destruction of SIP dialog '6994090221311200851-1168871725@172.16.100.100' in 32000 ms (Method: INVITE) localhost*CLI> <--- Reliably Transmitting (no NAT) to 172.16.100.100:5060 ---> SIP/2.0 486 Busy here Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e53-ac106464-3455392;received=172.16.100.100 From: ;tag=9750 To: ;tag=as578e547f Call-ID: 6994090221311200851-1168871725@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 15 15:35:27] DEBUG[24328]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10995 [Jan 15 15:35:27] DEBUG[24328]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-088bda18 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:27] DEBUG[24333]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0264249911243881@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200851-1168871725@172.16.100.100 From: ;tag=9750 To: ;tag=as578e547f CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e53-ac106464-3455392 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0264249911243881@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200851-1168871725@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=9750 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as578e547f (70) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e53-ac106464-3 455392 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10995 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200851-1168871725@172.16.100.100' of Response 1: Mat ch Not Found Really destroying SIP dialog '667a36a54544847e60a3666819395757@217.113.64.11' Method: INVITE [Jan 15 15:35:27] DEBUG[23173]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[22604]: chan_sip.c:5576 reqprep: Strict routing enforced for session 71097f0a2091398b11de350c29816d7a@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:2209709661@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK527df954;rport From: "972334913" ;tag=as0d58dbc6 To: ;tag=t1168870944-co738 Contact: Call-ID: 71097f0a2091398b11de350c29816d7a@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 15 15:35:27] DEBUG[22604]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10996 localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 180 Ringing From: "249912301506" ;tag=as19fa0939 To: ;tag=c733432d CSeq: 102 INVITE Call-ID: 08f411773d78de697c7d1ac41c2dc14a@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0a610aee;rport=5060 Contact: Content-Length: 151 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 10560248 10560249 IN IP4 10.10.13.4 s=Sip Call c=IN IP4 196.29.163.6 t=0 0 m=audio 40630 RTP/AVP 18 a=rtpmap:18 G729/8000 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "249912301506" ;tag=as19fa0939 (68) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=c733432d (48) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 08f411773d78de697c7d1ac41c2dc14a@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0a610aee;rport=5060 (69) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (56) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 151 (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=HuaweiSoftX3000 10560248 10560249 IN IP4 10.10.13.4 (53) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=Sip Call (10) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 196.29.163.6 (21) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40630 RTP/AVP 18 (24) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) --- (9 headers 7 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '08f411773d78de697c7d1ac 41c2dc14a@217.113.64.11' Request 102: Found [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 180 to standard invite [Jan 15 15:35:27] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/19 6.29.163.6-08c7db18 Found RTP audio format 18 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 196.29.163.6 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 196.29.163.6 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 196.29.163.6:40630 [Jan 15 15:35:27] DEBUG[16977]: channel.c:943 channel_find_locked: Avoiding initial deadlock for channel '0x8493688' Found description format G729 for ID 18 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/196.29.163.6-08c7db18 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 196.29.163.6:40630 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call -- SIP/196.29.163.6-08c7db18 is ringing <--- 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-6110000000498e4d-ac106464-3455390;received=172.16.100.100 From: ;tag=23863 To: ;tag=as3a9bfb4a Call-ID: 6994090221311200845-1168871720@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/196.29.163.6-08c7db18 is making progress passing it to SIP/5060-084d9978 <--- 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=z9hG4bK5a445744 To: ;tag=t1168871719-co200 From: "395459803620" ;tag=as564d41ab Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 Seq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO200-CPO02051 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1694188404 1694188404 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 40248 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 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK5a445744 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871719-co200 (57) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "395459803620" ;tag=as564d41ab (68) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO200-CPO02051 (60) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 230 (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 1694188404 1694188404 IN IP4 216.226.69.244 (47) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 40248 RTP/AVP 18 101 (28) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '658c624977ccfaab01016cd73d24d663@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:40248 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-0894c918 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 216.226.69.237:40248 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 658c624977ccfaab01016cd73d24d663@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:2205667373@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0dd6f2dc;rport From: "395459803620" ;tag=as564d41ab To: ;tag=t1168871719-co200 Contact: Call-ID: 658c624977ccfaab01016cd73d24d663@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:27] DEBUG[24313]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-0894c918 -- SIP/216.226.69.244-0894c918 answered SIP/5060-087eaf80 [Jan 15 15:35:27] DEBUG[24313]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-087eaf80 [Jan 15 15:35:27] DEBUG[24313]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-087eaf80 [Jan 15 15:35:27] DEBUG[24313]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:27] DEBUG[24313]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:27] DEBUG[24313]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 15718 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:27] DEBUG[24313]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:27] DEBUG[24313]: 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-6110000000498e49-ac106464-3455386;received=172.16.100.100 From: ;tag=22954 To: ;tag=as6354cf6c Call-ID: 6994090221311200841-1168871718@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 16974 16975 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 15718 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 15 15:35:27] DEBUG[24313]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10997 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432205667373@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200841-1168871718@172.16.100.100 From: ;tag=22954 To: ;tag=as6354cf6c CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e49-ac106464-3455386 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432205667373@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200841-1168871718@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=22954 (65) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as6354cf6c (67) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e49-ac106464-3 455386 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10997 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200841-1168871718@172.16.100.100' of Response 1: Mat ch Not Found [Jan 15 15:35:27] DEBUG[24313]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:27] DEBUG[24322]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:27] DEBUG[24322]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/196.29.163.6 - state 6 (Ringing) [Jan 15 15:35:27] DEBUG[24334]: app_queue.c:546 changethread: Device 'SIP/196.29.163.6' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:27] DEBUG[24335]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:27] DEBUG[24336]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. <--- SIP read from 172.16.100.100:5060 ---> INFO sip:0432209709661@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311199525-1168870943@172.16.100.100 From: ;tag=14931 To: ;tag=as12e8d479 CSeq: 3 INFO Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498925-ac106464-3455395 Supported: timer,100rel Max-Forwards: 70 Content-Length: 11 Content-Type: application/dtmf-relay Signal= # <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INFO sip:0432209709661@10.100.20.11:5060;user=phone SIP/2.0 (59) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311199525-1168870943@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=14931 (62) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as12e8d479 (67) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 3 INFO (12) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498925-ac106464-3 455395 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 11 (18) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Type: application/dtmf-relay (36) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: Signal= # (9) --- (10 headers 1 lines) --- [Jan 15 15:35:27] DEBUG[16996]: 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-6110000000498925-ac106464-3455395;received=172.16.100.100 From: ;tag=14931 To: ;tag=as12e8d479 Call-ID: 6994090221311199525-1168870943@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 <------------> <--- 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=z9hG4bK7de4aefb To: ;tag=t1168871714-co174 From: "4921199999" ;tag=as0028df42 Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 CANCEL User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO174-CPO01951 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7de4aefb (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871714-co174 (57) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0028df42 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 CANCEL (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO174-CPO01951 (60) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10992 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '63d06de74913b81e3da871cc28c5a258@217.113.64.11' of Request 102: M atch Not Found <--- 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=z9hG4bK7de4aefb To: ;tag=t1168871714-co174 From: "4921199999" ;tag=as0028df42 Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO174-CPO01951 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK7de4aefb (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871714-co174 (57) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "4921199999" ;tag=as0028df42 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO174-CPO01951 (60) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '63d06de74913b81e3da871cc28c5a258@217.113.64.11' of Request 102: M atch Found [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 216.226.69.244:5060: ACK sip:2204495604@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7de4aefb;rport From: "4921199999" ;tag=as0028df42 To: ;tag=t1168871714-co174 Contact: Call-ID: 63d06de74913b81e3da871cc28c5a258@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:27] DEBUG[24026]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[23176]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[22571]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[24182]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[23870]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[22636]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[20233]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[23844]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[24190]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes <--- 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=z9hG4bK43a6488b To: ;tag=t1168871707-co128 From: "441213288847" ;tag=as735c10a2 Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 CSeq: 102 INVITE Contact: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO128-CPO01818 Content-Type: application/sdp Content-Length: 228 v=0 o=- 717968811 717968811 IN IP4 216.226.69.244 s=ENS Session c=IN IP4 216.226.69.237 t=0 0 m=audio 43412 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 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK43a6488b (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168871707-co128 (57) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "441213288847" ;tag=as735c10a2 (68) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Contact: (45) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: User-Agent: ENS2.2.112-IS1-RMRG1001-RG575-EP0-CO128-CPO01818 (60) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Content-Length: 228 (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: (0) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=- 717968811 717968811 IN IP4 216.226.69.244 (45) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=ENS Session (13) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 216.226.69.237 (23) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 43412 RTP/AVP 18 101 (28) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) --- (10 headers 11 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2047 __sip_ack: Acked pending invite 102 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11' of Request 102: M atch Not Found [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 216.226.69.237:43412 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5062 process_sdp: T38 state changed to 0 on channel SIP/216.226.69.244-085f9540 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 216.226.69.237:43412 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5146 process_sdp: We have an owner, now see if we need to change this call [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: list_route: hop: [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5576 reqprep: Strict routing enforced for session 0ee407ed3ebf5f8e0137b11d4db58b5a@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:2204391307@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK20f3792c;rport From: "441213288847" ;tag=as735c10a2 To: ;tag=t1168871707-co128 Contact: Call-ID: 0ee407ed3ebf5f8e0137b11d4db58b5a@217.113.64.11 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:27] DEBUG[24280]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-085f9540 -- SIP/216.226.69.244-085f9540 answered SIP/5060-08b6fc60 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:27] DEBUG[24280]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08b6fc60 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:27] DEBUG[24280]: chan_sip.c:3420 sip_answer: SIP answering channel: SIP/5060-08b6fc60 [Jan 15 15:35:27] DEBUG[24337]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:27] DEBUG[24280]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:27] DEBUG[24280]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:27] DEBUG[24280]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 11842 [Jan 15 15:35:27] DEBUG[24338]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:27] DEBUG[24280]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:27] DEBUG[24280]: 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-6110000000498e27-ac106464-3455369;received=172.16.100.100 From: ;tag=1573 To: ;tag=as28e54cfd Call-ID: 6994090221311200807-1168871707@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 16974 16975 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 11842 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 15 15:35:27] DEBUG[24280]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #10999 <--- SIP read from 172.16.100.100:5060 ---> ACK sip:0432204391307@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200807-1168871707@172.16.100.100 From: ;tag=1573 To: ;tag=as28e54cfd CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e27-ac106464-3455369 Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: ACK sip:0432204391307@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200807-1168871707@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=1573 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as28e54cfd (67) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 1 ACK (11) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e27-ac106464-3 455369 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10999 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '6994090221311200807-1168871707@172.16.100.100' of Response 1: Mat ch Not Found [Jan 15 15:35:27] DEBUG[24280]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0432209930885@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200855-1168871727@172.16.100.100 From: ;tag=10659 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e57-ac106464-3455396 Contact: sip:34937124892@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 7405 7405 IN IP4 10.57.3.27 s=- c=IN IP4 10.57.3.27 t=0 0 m=audio 56332 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 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0432209930885@10.100.20.11:5060;user=phone SIP/2.0 (61) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200855-1168871727@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=10659 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (52) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e57-ac106464-3 455396 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:34937124892@172.16.100.100:5060;user=phone (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 7405 7405 IN IP4 10.57.3.27 (40) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.3.27 (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 56332 RTP/AVP 18 0 101 13 (33) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200855-1168871727@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:27] DEBUG[16996]: 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 - 6994090221311200855-1168871727@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:27] DEBUG[16996]: 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 15 15:35:27] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.3.27:56332 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 15 15:35:27] DEBUG[16996]: 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 10.57.3.27:56332 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0432209930885 in default (domain 10.100.20.11) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:7894 build_route: build_route: Contact hop: sip:34937124892@172.16.100.100:5060;user=phone list_route: hop: [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:13365 handle_request_invite: SIP/5060-085bef78: 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-6110000000498e57-ac106464-3455396;received=172.16.100.100 From: ;tag=10659 To: Call-ID: 6994090221311200855-1168871727@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 15 15:35:27] DEBUG[16996]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-085bef78 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:27] DEBUG[24339]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [0432209930885@default:1] Dial("SIP/5060-085bef78", "SIP/2209930885@216.226.69.244||t") in new stack [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:15155 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Jan 15 15:35:27] DEBUG[24340]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:3761 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:3763 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:3784 sip_new: This channel will not be able to handle video. [Jan 15 15:35:27] DEBUG[24339]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-default-0432209930885-1. [Jan 15 15:35:27] DEBUG[24339]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 15 15:35:27] DEBUG[24339]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 15 15:35:27] DEBUG[24339]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:2804 sip_call: Outgoing Call for 2209930885 [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:2818 sip_call: Our T38 capability (3872), joint T38 capability (3872) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 217.113.64.11 port 12224 Adding codec 0x100 (g729) to SDP [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:6298 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:2209930885@216.226.69.244 SIP/2.0 (44) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0cfa8558;rport (64) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 2: From: "34937124892" ;tag=as3036bb1a (66) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 3: To: (35) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 4: Contact: (40) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 5: Call-ID: 56b22f5b270c3bda7cb53dfa6267965f@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 7: User-Agent: gatewaycomms (24) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 9: Date: Mon, 15 Jan 2007 14:35:27 GMT (35) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 11: Supported: replaces (19) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 13: Content-Length: 209 (19) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4519 parse_request: Header 14: (0) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: o=root 16974 16974 IN IP4 217.113.64.11 (39) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: s=session (9) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: c=IN IP4 217.113.64.11 (22) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: m=audio 12224 RTP/AVP 18 (24) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: a=ptime:20 (10) [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:4551 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 216.226.69.244:5060: INVITE sip:2209930885@216.226.69.244 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0cfa8558;rport From: "34937124892" ;tag=as3036bb1a To: Contact: Call-ID: 56b22f5b270c3bda7cb53dfa6267965f@217.113.64.11 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Mon, 15 Jan 2007 14:35:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 209 v=0 o=root 16974 16974 IN IP4 217.113.64.11 s=session c=IN IP4 217.113.64.11 t=0 0 m=audio 12224 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 15 15:35:27] DEBUG[24339]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #11000 -- Called 2209930885@216.226.69.244 <--- SIP read from 172.16.100.100:5060 ---> INVITE sip:0264249915468232@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311200853-1168871726@172.16.100.100 From: ;tag=26589 o: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e55-ac106464-3455397 Contact: sip:2356648116@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 9655 9655 IN IP4 10.57.2.121 s=- c=IN IP4 10.57.2.121 t=0 0 m=audio 54012 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 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: INVITE sip:0264249915468232@10.100.20.11:5060;user=phone SIP/2.0 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311200853-1168871726@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=26589 (63) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Content-Type: application/sdp (29) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 1 INVITE (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498e55-ac106464-3 455397 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Contact: sip:2356648116@172.16.100.100:5060;user=phone (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: Supported: timer,100rel (23) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 11: Content-Length: 222 (19) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 12: (0) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: v=0 (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: o=MG4000|2.0 9655 9655 IN IP4 10.57.2.121 (41) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: s=- (3) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: c=IN IP4 10.57.2.121 (20) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: t=0 0 (5) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: m=audio 54012 RTP/AVP 18 101 13 (31) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=fmtp:18 annexb=yes (20) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=ptime:40 (10) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4551 parse_request: Line: a=rtpmap:13 CN/8000 (19) --- (12 headers 11 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2561 do_setnat: Setting NAT on UDPTL to Off [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4256 sip_alloc: Allocating new SIP dialog for 6994090221311200853-1168871726@172.16.100.100 - INVITE (With RT P) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1656 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -timer- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1670 parse_sip_options: Matched SIP option: timer [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1664 parse_sip_options: Found SIP option: -100rel- [Jan 15 15:35:27] DEBUG[16996]: 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 - 6994090221311200853-1168871726@172.16.100.100 Found peer 'VERAZ' [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2551 do_setnat: Setting NAT on RTP to Off [Jan 15 15:35:27] DEBUG[16996]: 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 15 15:35:27] DEBUG[16996]: chan_sip.c:4832 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.57.2.121:54012 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 15 15:35:27] DEBUG[16996]: 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 10.57.2.121:54012 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:5139 process_sdp: We're settling with these formats: 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:13290 handle_request_invite: Checking SIP call limits for device [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call Looking for 0264249915468232 in default (domain 10.100.20.11) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3758 sip_new: *** Our native formats are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3759 sip_new: *** Joint capabilities are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3760 sip_new: *** Our capabilities are 0x100 (g729) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:3761 sip_new: [Jan 15 15:35:27] DEBUG[22604]: chan_sip.c:5576 reqprep: Strict routing enforced for session 71 097f0a2091398b11de350c29816d7a@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:2209709661@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK40dd95c4;rport From: "972334913" ;tag=as0d58dbc6 To: ;tag=t1168870944-co738 Contact: Call-ID: 71097f0a2091398b11de350c29816d7a@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 15 15:35:27] DEBUG[22604]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #11004 [Jan 15 15:35:27] DEBUG[24044]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[20560]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes <--- 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=z9hG4bK527df954 To: ;tag=t1168870944-co738 From: "972334913" ;tag=as0d58dbc6 Call-ID: 71097f0a2091398b11de350c29816d7a@217.113.64.11 CSeq: 103 INFO User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO738-CPO00168 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK527df954 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168870944-co738 (57) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972334913" ;tag=as0d58dbc6 (62) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 71097f0a2091398b11de350c29816d7a@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 103 INFO (14) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO738-CPO00168 (60) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10996 [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '71097f0a2091398b11de350c29816d7a@217.113.64.11' of Request 103: M atch Not Found [Jan 15 15:35:27] DEBUG[23923]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:27] DEBUG[22317]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes localhost*CLI> <--- SIP read from 196.29.163.6:5060 ---> SIP/2.0 100 Trying From: "2356648116" ;tag=as17e47ca4 To: CSeq: 102 INVITE Call-ID: 51836176257c954235b2efd65fe353a3@217.113.64.11 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK712f0c4e;rport=5060 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: From: "2356648116" ;tag=as17e47ca4 (64) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: CSeq: 102 INVITE (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 51836176257c954235b2efd65fe353a3@217.113.64.11 (55) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK712f0c4e;rport=5060 (69) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #11002 - INVITE (got response) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '51836176257c954235b2efd 65fe353a3@217.113.64.11' Request 102: Found [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:27] DEBUG[22557]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes [Jan 15 15:35:27] DEBUG[24288]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> BYE sip:0432204464637@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311199497-1168870929@172.16.100.100 From: ;tag=8570 To: ;tag=as55a24fec CSeq: 2 BYE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498909-ac106464-3455398 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: BYE sip:0432204464637@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311199497-1168870929@172.16.100.100 (54) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=8570 (63) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as55a24fec (67) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 2 BYE (11) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498909-ac106464-3 455398 (85) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 172.16.100.100 : 5060 (no NAT) [Jan 15 15:35:27] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311199497-1168870929@172.16.100.100 [Jan 15 15:35:27] DEBUG[16996]: 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-6110000000498909-ac106464-3455398;received=172.16.100.100 From: ;tag=8570 To: ;tag=as55a24fec Call-ID: 6994090221311199497-1168870929@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 15 15:35:27] DEBUG[22557]: channel.c:3682 ast_generic_bridge: Didn't get a frame from channel: SIP/5060-08b3c210 [Jan 15 15:35:27] DEBUG[22557]: channel.c:3986 ast_channel_bridge: Bridge stops bridging channels SIP/5060-08b3c210 and SIP/216.226.69.244-08f88d00 [Jan 15 15:35:27] DEBUG[22557]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-08f88d00' [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-08f88d00, SIP callid 0f7cf3265344d8731438d68a41cb9ce7@217.113 .64.11) [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:3278 sip_hangup: update_call_counter(2204464637) - decrement call limit counter on hangup [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call Scheduling destruction of SIP dialog '0f7cf3265344d8731438d68a41cb9ce7@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:5576 reqprep: Strict routing enforced for session 0f7cf3265344d8731438d68a41cb9ce7@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:2204464637@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK7ce7d60d;rport From: "41765440890" ;tag=as34c33a6b To: ;tag=t1168870930-co725 Call-ID: 0f7cf3265344d8731438d68a41cb9ce7@217.113.64.11 CSeq: 103 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #11006 [Jan 15 15:35:27] DEBUG[22557]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-08f88d00 [Jan 15 15:35:27] DEBUG[22557]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:27] DEBUG[22557]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 15 15:35:27] DEBUG[22557]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,0432204464637,1) exited non-zero on 'SIP/5060-08b3c210' == Spawn extension (default, 0432204464637, 1) exited non-zero on 'SIP/5060-08b3c210' [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:27] DEBUG[22557]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:27] DEBUG[24343]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:27] DEBUG[22557]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-08b3c210' [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-08b3c210, SIP callid 6994090221311199497-1168870929@172.16.100.100) [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:27] DEBUG[22557]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:27] DEBUG[22557]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-08b3c210 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:27] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:27] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:27] DEBUG[24344]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:27] DEBUG[23992]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes Really destroying SIP dialog '6994090221311199497-1168870929@172.16.100.100' Method: BYE [Jan 15 15:35:27] DEBUG[22864]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:27] DEBUG[24210]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:28] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:28] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:28] DEBUG[24309]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:28] DEBUG[24319]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:28] DEBUG[24231]: 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=z9hG4bK0cfa8558 To: From: "34937124892" ;tag=as3036bb1a Call-ID: 56b22f5b270c3bda7cb53dfa6267965f@217.113.64.11 CSeq: 102 INVITE User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO240-CPO00000 Content-Length: 0 <-------------> [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK0cfa8558 (64) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: (35) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "34937124892" ;tag=as3036bb1a (66) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 56b22f5b270c3bda7cb53dfa6267965f@217.113.64.11 (55) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG0-RG575-EP0-CO240-CPO00000 (57) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:2098 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #11000 - INVITE (got response) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:2107 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '56b22f5b270c3bda7cb53df a6267965f@217.113.64.11' Request 102: Found [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:11539 handle_response_invite: SIP response 100 to standard invite [Jan 15 15:35:28] DEBUG[24214]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:28] DEBUG[24143]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:28] DEBUG[24339]: rtp.c:870 ast_rtcp_read: Got RTCP report of 40 bytes [Jan 15 15:35:28] DEBUG[24339]: chan_sip.c:6353 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 15 15:35:28] DEBUG[24339]: chan_sip.c:6121 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Jan 15 15:35:28] DEBUG[24339]: chan_sip.c:6122 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.100.20.11 port 10908 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jan 15 15:35:28] DEBUG[24339]: chan_sip.c:6253 add_sdp: -- Done with adding codecs to SDP [Jan 15 15:35:28] DEBUG[24339]: 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-6110000000498e57-ac106464-3455396;received=172.16.100.100 From: ;tag=10659 To: ;tag=as776ffb36 Call-ID: 6994090221311200855-1168871727@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 16974 16974 IN IP4 10.100.20.11 s=session c=IN IP4 10.100.20.11 t=0 0 m=audio 10908 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 15 15:35:28] DEBUG[24339]: rtp.c:2667 ast_rtp_write: Ooh, format changed from unknown to g729 [Jan 15 15:35:28] DEBUG[24339]: rtp.c:2684 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Jan 15 15:35:28] DEBUG[24231]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:28] DEBUG[22604]: 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=z9hG4bK40dd95c4 To: ;tag=t1168870944-co738 From: "972334913" ;tag=as0d58dbc6 Call-ID: 71097f0a2091398b11de350c29816d7a@217.113.64.11 CSeq: 104 INFO User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO738-CPO00168 Content-Length: 0 <-------------> [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.64.11:5060;rport;branch=z9hG4bK40dd95c4 (64) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: To: ;tag=t1168870944-co738 (57) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: From: "972334913" ;tag=as0d58dbc6 (62) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: Call-ID: 71097f0a2091398b11de350c29816d7a@217.113.64.11 (55) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: CSeq: 104 INFO (14) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: User-Agent: ENS2.2.112-IS1-RMRG2001-RG575-EP0-CO738-CPO00168 (60) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Content-Length: 0 (17) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:2055 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11004 [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:2065 __sip_ack: Stopping retransmission on '71097f0a2091398b11de350c29816d7a@217.113.64.11' of Request 104: M atch Not Found [Jan 15 15:35:28] DEBUG[22604]: rtp.c:870 ast_rtcp_read: Got RTCP report of 24 bytes localhost*CLI> <--- SIP read from 172.16.100.100:5060 ---> BYE sip:0432209709661@10.100.20.11:5060;user=phone SIP/2.0 Call-ID: 6994090221311199525-1168870943@172.16.100.100 From: ;tag=14931 To: ;tag=as12e8d479 CSeq: 4 BYE Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498925-ac106464-3455399 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 0: BYE sip:0432209709661@10.100.20.11:5060;user=phone SIP/2.0 (58) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 1: Call-ID: 6994090221311199525-1168870943@172.16.100.100 (54) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 2: From: ;tag=14931 (62) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 3: To: ;tag=as12e8d479 (67) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 4: CSeq: 4 BYE (11) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-6110000000498925-ac106464-3 455399 (85) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 6: Supported: timer,100rel (23) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 8: Content-Length: 0 (17) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:4519 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:14502 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 172.16.100.100 : 5060 (no NAT) [Jan 15 15:35:28] DEBUG[16996]: chan_sip.c:1609 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 6994090221311199525-1168870943@172.16.100.100 [Jan 15 15:35:28] DEBUG[16996]: 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-6110000000498925-ac106464-3455399;received=172.16.100.100 From: ;tag=14931 To: ;tag=as12e8d479 Call-ID: 6994090221311199525-1168870943@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 15 15:35:28] DEBUG[22604]: channel.c:3682 ast_generic_bridge: Didn't get a frame from channel: SIP/5060-089a10f8 [Jan 15 15:35:28] DEBUG[22604]: channel.c:3986 ast_channel_bridge: Bridge stops bridging channels SIP/5060-089a10f8 and SIP/216.226.69.244-085e3218 [Jan 15 15:35:28] DEBUG[22604]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/216.226.69.244-085e3218' [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:3270 sip_hangup: Hangup call SIP/216.226.69.244-085e3218, SIP callid 71097f0a2091398b11de350c29816d7a@217.113 .64.11) [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:3278 sip_hangup: update_call_counter(2209709661) - decrement call limit counter on hangup [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:2970 update_call_counter: Updating call counter for outgoing call Scheduling destruction of SIP dialog '71097f0a2091398b11de350c29816d7a@217.113.64.11' in 32000 ms (Method: INVITE) [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:5576 reqprep: Strict routing enforced for session 71097f0a2091398b11de350c29816d7a@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:2209709661@216.226.69.244:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.64.11:5060;branch=z9hG4bK0c08f0b6;rport From: "972334913" ;tag=as0d58dbc6 To: ;tag=t1168870944-co738 Call-ID: 71097f0a2091398b11de350c29816d7a@217.113.64.11 CSeq: 105 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:1951 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #11009 [Jan 15 15:35:28] DEBUG[22604]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/21 6.226.69.244-085e3218 [Jan 15 15:35:28] DEBUG[22604]: rtp.c:1473 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 15 15:35:28] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 216.226.69.244 [Jan 15 15:35:28] DEBUG[22604]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 15 15:35:28] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 216.226.69.244 [Jan 15 15:35:28] DEBUG[22604]: pbx.c:2363 __ast_pbx_run: Spawn extension (default,0432209709661,1) exited non-zero on 'SIP/5060-089a10f8' == Spawn extension (default, 0432209709661, 1) exited non-zero on 'SIP/5060-089a10f8' [Jan 15 15:35:28] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/216.226.69.244 - state 2 (In use) [Jan 15 15:35:28] DEBUG[22604]: cdr.c:896 ast_cdr_detach: Dropping CDR ! [Jan 15 15:35:28] DEBUG[24345]: app_queue.c:546 changethread: Device 'SIP/216.226.69.244' changed to state '2' (In use) but we don't care because they'r e not a member of any queue. [Jan 15 15:35:28] DEBUG[22604]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/5060-089a10f8' [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:3270 sip_hangup: Hangup call SIP/5060-089a10f8, SIP callid 6994090221311199525-1168870943@172.16.100.100) [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:3278 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Jan 15 15:35:28] DEBUG[22604]: chan_sip.c:2970 update_call_counter: Updating call counter for incoming call [Jan 15 15:35:28] DEBUG[22604]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/50 60-089a10f8 [Jan 15 15:35:28] DEBUG[16977]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060 [Jan 15 15:35:28] DEBUG[16977]: chan_sip.c:15097 sip_devicestate: Checking device state for peer 5060 [Jan 15 15:35:28] DEBUG[16977]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid) [Jan 15 15:35:28] DEBUG[24346]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a m ember of any queue. [Jan 15 15:35:28] DEBUG[24280]: rtp.c:870 ast_rtcp_read: Got RTCP report of 84 bytes [Jan 15 15:35:28] DEBUG[24286]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes Really destroying SIP dialog '6994090221311199525-1168870943@172.16.100.100' Method: BYE [Jan 15 15:35:28] DEBUG[23340]: rtp.c:870 ast_rtcp_read: Got RTCP report of 72 bytes [Jan 15 15:35:28] WARNING[16996]: chan_sip.c:1875 retrans_pkt: Maximum retries exceeded on transmission 6994090221311200803-1168871705@172.16.100.100 fo r seqno 1 (Critical Response) Really destroying SIP dialog '6994090221311200803-1168871705@172.16.100.100' Method: CANCEL localhost*CLI>