Script started on Tue 17 Oct 2006 08:39:13 BST root@asterisk1 asterisk]# /usr/sbin/asterisk -vvvvddddcn Asterisk SVN-trunk-r44241, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf <> Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'asterisk1' (pid 7378)*CLI> sip debug SIP Debugging enabled *CLI> <-- SIP read from 10.69.255.246:5060: INVITE sip:403@sipgate.co.uk:5060;transport=UDP SIP/2.0 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403" Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 1 INVITE Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535ae99-64f679b4 Max-Forwards: 70 Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Content-Type: application/sdp Content-Length: 193 v=0 o=rtp 1161071068 1161071068 IN IP4 10.69.255.246 s=- c=IN IP4 10.69.255.246 t=0 0 m=audio 5004 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 0: INVITE sip:403@sipgate.co.uk:5060;transport=UDP SIP/2.0 (55) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 1: From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f (94) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 2: To: "403" (37) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 3: Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 (72) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 4: CSeq: 1 INVITE (14) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 5: Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535ae99-64f679b4 (83) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 6: Max-Forwards: 70 (16) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 7: Supported: replaces (19) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 8: Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL (46) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 9: Contact: (54) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 10: Content-Type: application/sdp (29) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 11: Content-Length: 193 (19) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 12: (0) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: v=0 (3) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: o=rtp 1161071068 1161071068 IN IP4 10.69.255.246 (48) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: s=- (3) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: c=IN IP4 10.69.255.246 (22) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: t=0 0 (5) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: m=audio 5004 RTP/AVP 0 101 (26) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=fmtp:101 0-15 (15) --- (12 headers 9 lines) --- [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4163 sip_alloc: Allocating new SIP dialog for 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 - INVITE (With RTP) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:14175 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:1624 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:1632 parse_sip_options: Found SIP option: -replaces- [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:1638 parse_sip_options: Matched SIP option: replaces Sending to 10.69.255.246 : 5060 (NAT) Using INVITE request as basis request - 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:8760 check_user_full: Setting NAT on RTP to Off [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:8770 check_user_full: Setting NAT on UDPTL to Off Reliably Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535ae99-64f679b4;received=10.69.255.246 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403";tag=as4fbdd703 Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="inspiredbroadcast.net", nonce="3b7ee6b2" Content-Length: 0 --- [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:1915 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #14 Scheduling destruction of SIP dialog '100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246' in 32000 ms (Method: INVITE) Found user 'tulip1' <-- SIP read from 10.69.255.246:5060: ACK sip:403@sipgate.co.uk:5060;transport=UDP SIP/2.0 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403";tag=as4fbdd703 Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 1 ACK Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535ae99-64f679b4 Max-Forwards: 70 Contact: Content-Length: 0 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 0: ACK sip:403@sipgate.co.uk:5060;transport=UDP SIP/2.0 (52) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 1: From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f (94) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 2: To: "403";tag=as4fbdd703 (52) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 3: Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 (72) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 4: CSeq: 1 ACK (11) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 5: Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535ae99-64f679b4 (83) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 6: Max-Forwards: 70 (16) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 7: Contact: (54) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 8: Content-Length: 0 (17) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:14175 handle_request: **** Received ACK (6) - Command in SIP ACK [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:2014 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:2024 __sip_ack: Stopping retransmission on '100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246' of Response 1: Match Not Found <-- SIP read from 10.69.255.246:5060: INVITE sip:403@sipgate.co.uk:5060;transport=UDP SIP/2.0 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403" Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 2 INVITE Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535aecf-64492e8d Max-Forwards: 70 Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Proxy-Authorization: Digest username="tulip1",realm="inspiredbroadcast.net",nonce="3b7ee6b2",uri="sip:403@sipgate.co.uk:5060;transport=UDP",response="658fe03b4746fdf2c938621afd5d696b",algorithm=MD5 Content-Type: application/sdp Content-Length: 193 v=0 o=rtp 1161071068 1161071068 IN IP4 10.69.255.246 s=- c=IN IP4 10.69.255.246 t=0 0 m=audio 5004 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 0: INVITE sip:403@sipgate.co.uk:5060;transport=UDP SIP/2.0 (55) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 1: From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f (94) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 2: To: "403" (37) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 3: Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 (72) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 4: CSeq: 2 INVITE (14) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 5: Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535aecf-64492e8d (83) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 6: Max-Forwards: 70 (16) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 7: Supported: replaces (19) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 8: Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL (46) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 9: Contact: (54) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 10: Proxy-Authorization: Digest username="tulip1",realm="inspiredbroadcast.net",nonce="3b7ee6b2",uri="sip:403@sipgate.co.uk:5060;transport=UDP",response="658fe03b4746fdf2c938621afd5d696b",algorithm=MD5 (197) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 11: Content-Type: application/sdp (29) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 12: Content-Length: 193 (19) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 13: (0) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: v=0 (3) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: o=rtp 1161071068 1161071068 IN IP4 10.69.255.246 (48) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: s=- (3) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: c=IN IP4 10.69.255.246 (22) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: t=0 0 (5) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: m=audio 5004 RTP/AVP 0 101 (26) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=fmtp:101 0-15 (15) --- (13 headers 9 lines) --- [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:14175 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 10.69.255.246 : 5060 (NAT) Using INVITE request as basis request - 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:8760 check_user_full: Setting NAT on RTP to Off [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:8770 check_user_full: Setting NAT on UDPTL to Off Found user 'tulip1' Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.69.255.246:5004 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4705 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4917 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.69.255.246:5004 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4994 process_sdp: We're settling with these formats: 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:12970 handle_request_invite: Checking SIP call limits for device tulip1 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:2890 update_call_counter: Updating call counter for incoming call Looking for 403 in from-sip (domain sipgate.co.uk) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:3660 sip_new: *** Our native formats are 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:3661 sip_new: *** Joint capabilities are 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:3662 sip_new: *** Our capabilities are 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:3663 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:3686 sip_new: This channel will not be able to handle video. [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:7690 build_route: build_route: Contact hop: list_route: hop: [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:13041 handle_request_invite: SIP/tulip1-09844548: New call is still down.... Trying... Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535aecf-64492e8d;received=10.69.255.246 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403" Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Oct 17 08:39:31] DEBUG[7385]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/tulip1-09844548 [Oct 17 08:39:31] DEBUG[7381]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - tulip1 [Oct 17 08:39:31] DEBUG[7381]: chan_sip.c:14764 sip_devicestate: Checking device state for peer tulip1 [Oct 17 08:39:31] DEBUG[7381]: devicestate.c:287 do_state_change: Changing state for SIP/tulip1 - state 1 (Not in use) [Oct 17 08:39:31] DEBUG[7386]: pbx.c:1688 pbx_extension_helper: Launching 'Macro' -- Executing [403@from-sip:1] Macro("SIP/tulip1-09844548", "ext|SIP/tulip3") in new stack [Oct 17 08:39:31] DEBUG[7386]: pbx.c:1688 pbx_extension_helper: Launching 'Dial' -- Executing [s@macro-ext:1] Dial("SIP/tulip1-09844548", "SIP/tulip3|15") in new stack [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:14822 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4163 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:2519 create_addr_from_peer: Our T38 capability (3840) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:2529 create_addr_from_peer: Setting NAT on RTP to Off [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:2543 create_addr_from_peer: Setting NAT on UDPTL to Off [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:3660 sip_new: *** Our native formats are 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:3661 sip_new: *** Joint capabilities are 0x0 (nothing) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:3662 sip_new: *** Our capabilities are 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:3663 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:3665 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:3686 sip_new: This channel will not be able to handle video. [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:16374 sip_set_rtp_peer: Early remote bridge setting SIP '5cbf27d17491e92a2169297d7d83337b@10.69.255.251' - Sending media to 10.69.255.246 [Oct 17 08:39:31] DEBUG[7386]: rtp.c:1579 ast_rtp_make_compatible: Seeded SDP of 'SIP/tulip3-09856ea0' with that of 'SIP/tulip1-09844548' [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable STACK-macro-ext-s-1. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable ARG1. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable MACRO_PRIORITY. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable MACRO_CONTEXT. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable MACRO_EXTEN. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable STACK-from-sip-403-1. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Oct 17 08:39:31] DEBUG[7386]: channel.c:3151 ast_channel_inherit_variables: Not copying variable SIPURI. [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:2730 sip_call: Outgoing Call for tulip3 [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:2890 update_call_counter: Updating call counter for outgoing call [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:2738 sip_call: Our T38 capability (3840), joint T38 capability (3840) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:5903 add_sdp: ** Our capability: 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc) Video flag: False [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:5904 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 10.69.255.251 port 26588 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x20 (adpcm) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:6056 add_sdp: -- Done with adding codecs to SDP [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:6095 add_sdp: Done building SDP. Settling with this capability: 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 0: INVITE sip:tulip3@10.69.255.242:5060 SIP/2.0 (44) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 1: Via: SIP/2.0/UDP 10.69.255.251:5060;branch=z9hG4bK4ed719ff;rport (64) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 2: From: "Tulip ATA port 1" ;tag=as06b103e4 (63) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 3: To: (35) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 4: Contact: (32) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 5: Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 (55) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 6: CSeq: 102 INVITE (16) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 8: Max-Forwards: 70 (16) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 9: Date: Tue, 17 Oct 2006 07:39:31 GMT (35) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 11: Supported: replaces (19) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 12: Content-Type: application/sdp (29) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 13: Content-Length: 510 (19) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4383 parse_request: Header 14: (0) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: v=0 (3) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: o=root 7378 7378 IN IP4 10.69.255.246 (37) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: s=session (9) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: c=IN IP4 10.69.255.246 (22) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: t=0 0 (5) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: m=audio 5004 RTP/AVP 0 8 3 111 97 7 110 5 101 (45) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=fmtp:97 mode=30 (17) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:7 LPC/8000 (19) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:110 speex/8000 (23) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:5 DVI4/8000 (20) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=ptime:20 (10) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=fmtp:101 0-16 (15) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=silenceSupp:off - - - - (25) [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:4415 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.69.255.242:5060: INVITE sip:tulip3@10.69.255.242:5060 SIP/2.0 Via: SIP/2.0/UDP 10.69.255.251:5060;branch=z9hG4bK4ed719ff;rport From: "Tulip ATA port 1" ;tag=as06b103e4 To: Contact: Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 17 Oct 2006 07:39:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 510 v=0 o=root 7378 7378 IN IP4 10.69.255.246 s=session c=IN IP4 10.69.255.246 t=0 0 m=audio 5004 RTP/AVP 0 8 3 111 97 7 110 5 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:111 G726-32/8000 a=ptime:20 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:7 LPC/8000 a=ptime:20 a=rtpmap:110 speex/8000 a=ptime:20 a=rtpmap:5 DVI4/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- [Oct 17 08:39:31] DEBUG[7386]: chan_sip.c:1915 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #16 -- Called tulip3 <-- SIP read from 10.69.255.242:5060: SIP/2.0 180 Ringing From: "Tulip ATA port 1";tag=as06b103e4 To: ;tag=100c2b08-f2ff450a-13c4-453488b1-17b6b31c-453488b1 Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.69.255.251:5060;rport=5060;branch=z9hG4bK4ed719ff Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Content-Length: 0 [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 1: From: "Tulip ATA port 1";tag=as06b103e4 (62) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 2: To: ;tag=100c2b08-f2ff450a-13c4-453488b1-17b6b31c-453488b1 (89) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 3: Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 (55) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 4: CSeq: 102 INVITE (16) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 5: Via: SIP/2.0/UDP 10.69.255.251:5060;rport=5060;branch=z9hG4bK4ed719ff (69) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 6: Supported: replaces (19) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 7: Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL (46) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 8: Contact: (40) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 9: Content-Length: 0 (17) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:2057 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #16 - INVITE (got response) [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:2066 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5cbf27d17491e92a2169297d7d83337b@10.69.255.251' Request 102: Found [Oct 17 08:39:31] DEBUG[7385]: chan_sip.c:11308 handle_response_invite: SIP response 180 to standard invite [Oct 17 08:39:31] DEBUG[7385]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/tulip3-09856ea0 -- SIP/tulip3-09856ea0 is ringing [Oct 17 08:39:31] DEBUG[7386]: rtp.c:1514 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/tulip1-09844548' with that of 'SIP/tulip3-09856ea0' Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535aecf-64492e8d;received=10.69.255.246 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403";tag=as65214d66 Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Oct 17 08:39:31] DEBUG[7381]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - tulip3 [Oct 17 08:39:31] DEBUG[7381]: chan_sip.c:14764 sip_devicestate: Checking device state for peer tulip3 [Oct 17 08:39:31] DEBUG[7381]: devicestate.c:287 do_state_change: Changing state for SIP/tulip3 - state 1 (Not in use) <-- SIP read from 10.69.255.242:5060: SIP/2.0 200 OK From: "Tulip ATA port 1";tag=as06b103e4 To: ;tag=100c2b08-f2ff450a-13c4-453488b1-17b6b31c-453488b1 Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.69.255.251:5060;rport=5060;branch=z9hG4bK4ed719ff Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Content-Type: application/sdp Content-Length: 196 v=0 o=rtp 1161071428 1161071428 IN IP4 10.69.255.242 s=- c=IN IP4 10.69.255.242 t=0 0 m=audio 5004 RTP/AVP 2 101 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 0: SIP/2.0 200 OK (14) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 1: From: "Tulip ATA port 1";tag=as06b103e4 (62) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 2: To: ;tag=100c2b08-f2ff450a-13c4-453488b1-17b6b31c-453488b1 (89) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 3: Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 (55) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 4: CSeq: 102 INVITE (16) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 5: Via: SIP/2.0/UDP 10.69.255.251:5060;rport=5060;branch=z9hG4bK4ed719ff (69) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 6: Supported: replaces (19) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 7: Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL (46) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 8: Contact: (40) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 9: Content-Type: application/sdp (29) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 10: Content-Length: 196 (19) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 11: (0) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: v=0 (3) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: o=rtp 1161071428 1161071428 IN IP4 10.69.255.242 (48) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: s=- (3) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: c=IN IP4 10.69.255.242 (22) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: t=0 0 (5) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: m=audio 5004 RTP/AVP 2 101 (26) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=rtpmap:2 g726-32/8000 (23) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4415 parse_request: Line: a=fmtp:101 0-15 (15) --- (11 headers 9 lines) --- [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:2006 __sip_ack: Acked pending invite 102 [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:2024 __sip_ack: Stopping retransmission on '5cbf27d17491e92a2169297d7d83337b@10.69.255.251' of Request 102: Match Not Found [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:11308 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 10.69.255.242:5004 [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4705 process_sdp: Peer doesn't provide T.38 UDPTL Found description format g726-32 for ID 2 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4917 process_sdp: T38 state changed to 0 on channel SIP/tulip3-09856ea0 Capabilities: us - 0xeae (gsm|ulaw|alaw|g726|adpcm|lpc10|speex|ilbc), peer - audio=0x800 (g726)/video=0x0 (nothing), combined - 0x800 (g726) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.69.255.242:5004 [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4994 process_sdp: We're settling with these formats: 0x800 (g726) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:5001 process_sdp: We have an owner, now see if we need to change this call [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:5006 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0x800 (g726) and not 0x4 (ulaw) [Oct 17 08:39:39] DEBUG[7385]: channel.c:2684 set_format: Set channel SIP/tulip3-09856ea0 to read format ulaw [Oct 17 08:39:39] DEBUG[7385]: channel.c:2684 set_format: Set channel SIP/tulip3-09856ea0 to write format ulaw [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:2890 update_call_counter: Updating call counter for outgoing call [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:7690 build_route: build_route: Contact hop: list_route: hop: [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:5425 reqprep: Strict routing enforced for session 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 set_destination: Parsing for address/port to send to set_destination: set destination to 10.69.255.242, port 5060 Transmitting (no NAT) to 10.69.255.242:5060: ACK sip:tulip3@10.69.255.242:5060 SIP/2.0 Via: SIP/2.0/UDP 10.69.255.251:5060;branch=z9hG4bK7537b707;rport From: "Tulip ATA port 1" ;tag=as06b103e4 To: ;tag=100c2b08-f2ff450a-13c4-453488b1-17b6b31c-453488b1 Contact: Call-ID: 5cbf27d17491e92a2169297d7d83337b@10.69.255.251 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Oct 17 08:39:39] DEBUG[7386]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/tulip3-09856ea0 -- SIP/tulip3-09856ea0 answered SIP/tulip1-09844548 [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:16374 sip_set_rtp_peer: Early remote bridge setting SIP '100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246' - Sending media to 10.69.255.242 [Oct 17 08:39:39] DEBUG[7386]: rtp.c:1514 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/tulip1-09844548' with that of 'SIP/tulip3-09856ea0' [Oct 17 08:39:39] DEBUG[7386]: channel.c:2684 set_format: Set channel SIP/tulip1-09844548 to read format slin [Oct 17 08:39:39] DEBUG[7386]: channel.c:2684 set_format: Set channel SIP/tulip3-09856ea0 to write format slin [Oct 17 08:39:39] DEBUG[7386]: channel.c:2684 set_format: Set channel SIP/tulip3-09856ea0 to read format slin [Oct 17 08:39:39] DEBUG[7386]: channel.c:2684 set_format: Set channel SIP/tulip1-09844548 to write format slin [Oct 17 08:39:39] DEBUG[7386]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/tulip1-09844548 [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:3342 sip_answer: SIP answering channel: SIP/tulip1-09844548 [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:6150 transmit_response_with_sdp: Setting framing from config on incoming call [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:5903 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:5904 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.69.255.251 port 18738 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:6056 add_sdp: -- Done with adding codecs to SDP [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:6095 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) Reliably Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4534889f-5535aecf-64492e8d;received=10.69.255.246 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403";tag=as65214d66 Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 239 v=0 o=root 7378 7378 IN IP4 10.69.255.242 s=session c=IN IP4 10.69.255.242 t=0 0 m=audio 5004 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- [Oct 17 08:39:39] DEBUG[7386]: chan_sip.c:1915 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #18 [Oct 17 08:39:39] DEBUG[7386]: rtp.c:3161 ast_rtp_bridge: Channel codec0 = 4 is not codec1 = 2048, cannot native bridge in RTP. [Oct 17 08:39:39] DEBUG[7381]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - tulip3 [Oct 17 08:39:39] DEBUG[7381]: chan_sip.c:14764 sip_devicestate: Checking device state for peer tulip3 [Oct 17 08:39:39] DEBUG[7381]: devicestate.c:287 do_state_change: Changing state for SIP/tulip3 - state 1 (Not in use) [Oct 17 08:39:39] DEBUG[7381]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - tulip1 [Oct 17 08:39:39] DEBUG[7381]: chan_sip.c:14764 sip_devicestate: Checking device state for peer tulip1 [Oct 17 08:39:39] DEBUG[7381]: devicestate.c:287 do_state_change: Changing state for SIP/tulip1 - state 1 (Not in use) <-- SIP read from 10.69.255.246:5060: ACK sip:403@10.69.255.251 SIP/2.0 From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f To: "403";tag=as65214d66 Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 CSeq: 2 ACK Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-453488a7-5535cdce-f2c2e92 Max-Forwards: 70 Contact: Proxy-Authorization: Digest username="tulip1",realm="inspiredbroadcast.net",nonce="3b7ee6b2",uri="sip:403@sipgate.co.uk:5060;transport=UDP",response="658fe03b4746fdf2c938621afd5d696b",algorithm=MD5 Content-Length: 0 [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 0: ACK sip:403@10.69.255.251 SIP/2.0 (33) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 1: From: "Line 1";tag=100c37b0-f6ff450a-13c4-4534889f-57a308f7-4534889f (94) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 2: To: "403";tag=as65214d66 (52) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 3: Call-ID: 100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246 (72) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 4: CSeq: 2 ACK (11) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 5: Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-453488a7-5535cdce-f2c2e92 (82) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 6: Max-Forwards: 70 (16) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 7: Contact: (54) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 8: Proxy-Authorization: Digest username="tulip1",realm="inspiredbroadcast.net",nonce="3b7ee6b2",uri="sip:403@sipgate.co.uk:5060;transport=UDP",response="658fe03b4746fdf2c938621afd5d696b",algorithm=MD5 (197) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 9: Content-Length: 0 (17) [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:4383 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:14175 handle_request: **** Received ACK (6) - Command in SIP ACK [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:2014 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Oct 17 08:39:39] DEBUG[7385]: chan_sip.c:2024 __sip_ack: Stopping retransmission on '100c5be0-f6ff450a-13c4-4534889f-5b88f6b6-4534889f@10.69.255.246' of Response 2: Match Not Found =========== at this point call was set up but with audio not working; =========== the UDP streams used different codecs in both directions =========== No further messages until call hung up (snipped)