Asterisk 1.2.17, 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. ========================================================================= Connected to Asterisk 1.2.17 currently running on fargo (pid = 20784) fargo*CLI> Core debug is at least 4 fargo*CLI> <-- SIP read from 10.50.103.51:1036: INVITE sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK140f8b4617a3b5096 Max-Forwards: 70 From: Robert Arritt ;tag=f44e6a847d To: 42512 Call-ID: 6d577e66a409f163 CSeq: 30106 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt User-Agent: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 (50) fargo*CLI> Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK140f8b4617a3b5096 (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: Robert Arritt ;tag=f44e6a847d (72) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: 42512 (47) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 30106 INVITE (18) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk, hold, conference (36) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Contact: Robert Arritt (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: User-Agent: Aastra 55i/2.1.0.2067 (33) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 568 (19) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 0 IN IP4 10.50.103.51 (31) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 (67) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:106 BV16/8000 (22) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:107 BV32/16000 (23) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:113 L16/16000 (22) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:110 PCMU/16000 (23) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 PCMA/16000 (23) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:112 L16/8000 (21) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:115 G726-32/8000 (25) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:9 G722/8000 (20) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (13 headers 24 lines) --- Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3191 sip_alloc: Allocating new SIP dialog for 6d577e66a409f163 - INVITE (With RTP) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 6d577e66a409f163 Sending to 10.50.103.51 : 5060 (non-NAT) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK140f8b4617a3b5096;received=10.50.103.51 From: Robert Arritt ;tag=f44e6a847d To: 42512 ;tag=as2a86d3b0 Call-ID: 6d577e66a409f163 CSeq: 30106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4579e42d" Content-Length: 0 --- Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #21 Scheduling destruction of call '6d577e66a409f163' in 15000 ms Found user '42511' fargo*CLI> <-- SIP read from 10.50.103.51:1036: ACK sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK140f8b4617a3b5096 Max-Forwards: 70 From: Robert Arritt ;tag=f44e6a847d To: 42512 ;tag=as2a86d3b0 Call-ID: 6d577e66a409f163 CSeq: 30106 ACK User-Agent: Aastra 55i/2.1.0.2067 Content-Length: 0 Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: ACK sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 (47) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK140f8b4617a3b5096 (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) fargo*CLI> Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: Robert Arritt ;tag=f44e6a847d (72) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: 42512 ;tag=as2a86d3b0 (62) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 30106 ACK (15) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 55i/2.1.0.2067 (33) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Length: 0 (17) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: (0) --- (9 headers 0 lines) --- Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received ACK (6) - Command in SIP ACK Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Response 30106: Match Found fargo*CLI> <-- SIP read from 10.50.103.51:1036: INVITE sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK2450e34b685f3d480 Proxy-Authorization: Digest username="42511",realm="asterisk",nonce="4579e42d",uri="sip:42512@fargo.ana.aastra.com:6050",response="59e451d69e11613fba2fcaad5470dc1d",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=f44e6a847d To: 42512 Call-ID: 6d577e66a409f163 CSeq: 30107 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt User-Agent: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42512@fargo.ana.aastra.com:6050 SIP/2.0 (50) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK2450e34b685f3d480 (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Proxy-Authorization: Digest username="42511",realm="asterisk",nonce="4579e42d",uri="sip:42512@fargo.ana.aastra.com:6050",response="59e451d69e11613fba2fcaad5470dc1d",algorithm=MD5 (178) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: Max-Forwards: 70 (16) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: From: Robert Arritt ;tag=f44e6a847d (72) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: To: 42512 (47) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: CSeq: 30107 INVITE (18) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Allow-Events: talk, hold, conference (36) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Contact: Robert Arritt (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: User-Agent: Aastra 55i/2.1.0.2067 (33) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Type: application/sdp (29) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: Content-Length: 568 (19) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 14: (0) fargo*CLI> Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 0 IN IP4 10.50.103.51 (31) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 (67) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:106 BV16/8000 (22) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:107 BV32/16000 (23) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:113 L16/16000 (22) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:110 PCMU/16000 (23) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 PCMA/16000 (23) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:112 L16/8000 (21) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:115 G726-32/8000 (25) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:9 G722/8000 (20) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (14 headers 24 lines) --- Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 6d577e66a409f163 Sending to 10.50.103.51 : 5060 (non-NAT) fargo*CLI> Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Found user '42511' Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format G722 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x45c (ulaw|alaw|g726|slin|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:10669 handle_request_invite: Checking SIP call limits for device 42511 Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:2228 update_call_counter: Updating call counter for incoming call Looking for 42512 in default (domain fargo.ana.aastra.com) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:6267 build_route: build_route: Contact hop: Robert Arritt list_route: hop: Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK2450e34b685f3d480;received=10.50.103.51 From: Robert Arritt ;tag=f44e6a847d To: 42512 Call-ID: 6d577e66a409f163 CSeq: 30107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Mar 27 11:59:27 DEBUG[20813]: pbx.c:1697 pbx_extension_helper: Launching 'Dial' Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3191 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 Mar 27 11:59:27 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable STACK-default-42512-1. Mar 27 11:59:27 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPCALLID. Mar 27 11:59:27 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Mar 27 11:59:27 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Mar 27 11:59:27 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPURI. Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:2083 sip_call: Outgoing Call for 42512 Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call We're at 10.50.103.10 port 17432 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 (56) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2643f456;rport (63) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: (47) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 102 INVITE (16) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Date: Tue, 27 Mar 2007 15:59:27 GMT (35) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 263 (19) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20784 IN IP4 10.50.103.10 (38) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.10 (21) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 17432 RTP/AVP 0 8 3 101 (31) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 12 lines fargo*CLI> Mar 27 11:59:27 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42511 Mar 27 11:59:27 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42511 - state 2 (In use) Mar 27 11:59:27 DEBUG[20814]: app_queue.c:500 changethread: Device 'SIP/42511' changed to state '2' (In use) but we don't care because they're not a member of any queue. Mar 27 11:59:27 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42511 fargo*CLI> Reliably Transmitting (no NAT) to 10.50.103.54:5060: INVITE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2643f456;rport From: "Robert Arritt" ;tag=as1d7ac186 To: Contact: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Mar 2007 15:59:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 20784 20784 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 17432 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> Mar 27 11:59:27 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #23 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2643f456;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Server: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 180 Ringing (19) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2643f456;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) fargo*CLI> Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 102 INVITE (16) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Allow-Events: talk, hold, conference (36) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Contact: Robert Arritt (66) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Server: Aastra 35i/20070327 (27) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Content-Length: 0 (17) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: (0) --- (11 headers 0 lines) --- Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:1459 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #23 - INVITE (got response) Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' Request 102: Found Mar 27 11:59:27 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 180 to standard invite Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK2450e34b685f3d480;received=10.50.103.51 From: Robert Arritt ;tag=f44e6a847d To: 42512 ;tag=as24d4e1f6 Call-ID: 6d577e66a409f163 CSeq: 30107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Mar 27 11:59:27 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 Mar 27 11:59:27 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42512 - state 6 (Ringing) Mar 27 11:59:27 DEBUG[20815]: app_queue.c:500 changethread: Device 'SIP/42512' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Mar 27 11:59:27 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2643f456;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 102 INVITE Contact: Robert Arritt Server: Aastra 35i/20070327 Supported: timer Content-Type: application/sdp Content-Length: 256 v=0 o=MxSIP 0 0 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK2643f456;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 102 INVITE (16) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 35i/20070327 (27) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Supported: timer (16) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Type: application/sdp (29) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Content-Length: 256 (19) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: (0) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 0 IN IP4 10.50.103.54 (31) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) fargo*CLI> Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (11 headers 13 lines) --- Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' of Request 102: Match Found Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3000 Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.54:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:28 DEBUG[20800]: chan_sip.c:6267 build_route: build_route: Contact hop: Robert Arritt list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Transmitting (no NAT) to 10.50.103.54:5060: ACK sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3d069a59;rport From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Contact: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:2570 sip_answer: sip_answer(SIP/42511-081c5d68) We're at 10.50.103.10 port 17328 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bK2450e34b685f3d480;received=10.50.103.51 From: Robert Arritt ;tag=f44e6a847d To: 42512 ;tag=as24d4e1f6 Call-ID: 6d577e66a409f163 CSeq: 30107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 20784 20784 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 17328 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #24 fargo*CLI> Mar 27 11:59:28 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 Mar 27 11:59:28 DEBUG[20787]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/42512-081cc328' fargo*CLI> Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:13133 sip_set_rtp_peer: Deferring reinvite on SIP '6d577e66a409f163' - It's audio will be redirected to IP 10.50.103.54 Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:13127 sip_set_rtp_peer: Sending reinvite on SIP '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' - It's audio soon redirected to IP 10.50.103.51 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 We're at 10.50.103.10 port 17432 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 (56) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4be83ef5;rport (63) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 103 INVITE (16) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 321 (19) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20785 IN IP4 10.50.103.51 (38) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) fargo*CLI> Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 111 10 97 101 (38) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:10 L16/8000 (20) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 14 lines Reliably Transmitting (no NAT) to 10.50.103.54:5060: INVITE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4be83ef5;rport From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Contact: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 321 v=0 o=root 20784 20785 IN IP4 10.50.103.51 s=session c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:28 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #25 Mar 27 11:59:28 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42512 - state 2 (In use) fargo*CLI> Mar 27 11:59:28 DEBUG[20816]: app_queue.c:500 changethread: Device 'SIP/42512' changed to state '2' (In use) but we don't care because they're not a member of any queue. Mar 27 11:59:28 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 Mar 27 11:59:28 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42511 Mar 27 11:59:28 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42511 - state 2 (In use) Mar 27 11:59:28 DEBUG[20817]: app_queue.c:500 changethread: Device 'SIP/42511' changed to state '2' (In use) but we don't care because they're not a member of any queue. Mar 27 11:59:28 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42511 fargo*CLI> Mar 27 11:59:28 DEBUG[20813]: rtp.c:1361 ast_rtp_write: Ooh, format changed from unknown to ulaw fargo*CLI> Mar 27 11:59:28 DEBUG[20813]: rtp.c:411 ast_rtcp_read: Got RTCP report of 40 bytes fargo*CLI> Mar 27 11:59:29 DEBUG[20813]: rtp.c:1361 ast_rtp_write: Ooh, format changed from unknown to ulaw fargo*CLI> <-- SIP read from 10.50.103.51:1036: ACK sip:42512@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKf3b2a65fa2b32dcb7 Proxy-Authorization: Digest username="42511",realm="asterisk",nonce="4579e42d",uri="sip:42512@10.50.103.10:6050",response="41e94a818bd81ddf04da77dabee1920d",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=f44e6a847d To: 42512 ;tag=as24d4e1f6 Call-ID: 6d577e66a409f163 CSeq: 30107 ACK User-Agent: Aastra 55i/2.1.0.2067 Content-Length: 0 fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: ACK sip:42512@10.50.103.10:6050 SIP/2.0 (39) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.51:5060;branch=z9hG4bKf3b2a65fa2b32dcb7 (66) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Proxy-Authorization: Digest username="42511",realm="asterisk",nonce="4579e42d",uri="sip:42512@10.50.103.10:6050",response="41e94a818bd81ddf04da77dabee1920d",algorithm=MD5 (170) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: Max-Forwards: 70 (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: From: Robert Arritt ;tag=f44e6a847d (72) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: To: 42512 ;tag=as24d4e1f6 (62) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: CSeq: 30107 ACK (15) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: User-Agent: Aastra 55i/2.1.0.2067 (33) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Length: 0 (17) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) --- (10 headers 0 lines) --- Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received ACK (6) - Command in SIP ACK Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #24 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Response 30107: Match Found Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:9703 check_pendings: Sending pending reinvite on '6d577e66a409f163' set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 17328 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42511@10.50.103.51:5060 SIP/2.0 (42) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK51d6f69e;rport (63) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 102 INVITE (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 239 (19) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=root 20784 20785 IN IP4 10.50.103.54 (38) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=session (9) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) fargo*CLI> 13 headers, 11 lines fargo*CLI> Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK51d6f69e;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 20784 20785 IN IP4 10.50.103.54 s=session c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #26 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4be83ef5;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 103 INVITE Server: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 100 Trying (18) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4be83ef5;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 103 INVITE (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Server: Aastra 35i/20070327 (27) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Content-Length: 0 (17) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1459 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #25 - INVITE (got response) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' Request 103: Found Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 100 to RE-invite on outgoing call 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4be83ef5;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 103 INVITE Contact: Robert Arritt Server: Aastra 35i/20070327 Supported: timer Content-Type: application/sdp Content-Length: 316 v=0 o=MxSIP 0 12022 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4be83ef5;rport=6050;received=10.50.103.10 (90) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 103 INVITE (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 35i/20070327 (27) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Supported: timer (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Type: application/sdp (29) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Content-Length: 316 (19) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: (0) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 12022 IN IP4 10.50.103.54 (35) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 111 10 101 (35) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:10 L16/8000 (20) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) fargo*CLI> --- (11 headers 15 lines) --- Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 103 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' of Request 103: Match Found Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 10 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3000 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.54:3000 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format L16 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x5c (ulaw|alaw|g726|slin)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:6210 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Transmitting (no NAT) to 10.50.103.54:5060: ACK sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK19d358cf;rport From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Contact: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Mar 27 11:59:29 DEBUG[20813]: rtp.c:1676 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' changed end address to 10.50.103.54:3000 (format 92) Mar 27 11:59:29 DEBUG[20813]: rtp.c:1678 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' changed end vaddress to 0.0.0.0:0 (format 92) Mar 27 11:59:29 DEBUG[20813]: rtp.c:1680 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' was 10.50.103.54:3000/(format 12) Mar 27 11:59:29 DEBUG[20813]: rtp.c:1682 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' was 0.0.0.0:0/(format 12) Mar 27 11:59:29 DEBUG[20813]: chan_sip.c:13133 sip_set_rtp_peer: Deferring reinvite on SIP '6d577e66a409f163' - It's audio will be redirected to IP 10.50.103.54 fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK51d6f69e;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 102 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 26920 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK51d6f69e;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 102 INVITE (16) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Type: application/sdp (29) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Length: 260 (19) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 26920 IN IP4 10.50.103.51 (35) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (10 headers 13 lines) --- Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #26 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Request 102: Match Found Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6d577e66a409f163 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event fargo*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) fargo*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:6267 build_route: build_route: Contact hop: Robert Arritt fargo*CLI> list_route: hop: fargo*CLI> set_destination: Parsing for address/port to send to fargo*CLI> set_destination: set destination to 10.50.103.51, port 5060 fargo*CLI> Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK08045f91;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:9703 check_pendings: Sending pending reinvite on '6d577e66a409f163' fargo*CLI> set_destination: Parsing for address/port to send to fargo*CLI> set_destination: set destination to 10.50.103.51, port 5060 fargo*CLI> We're at 10.50.103.10 port 17328 fargo*CLI> Adding codec 0x4 (ulaw) to SDP fargo*CLI> Adding codec 0x8 (alaw) to SDP fargo*CLI> Adding codec 0x10 (g726) to SDP fargo*CLI> Adding codec 0x40 (slin) to SDP fargo*CLI> Adding non-codec 0x1 (telephone-event) to SDP fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 (56) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK54dcd0a2;rport (63) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 103 INVITE (16) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 295 (19) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: (0) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=root 20784 20786 IN IP4 10.50.103.54 (38) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=session (9) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 111 10 101 (35) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:10 L16/8000 (20) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) fargo*CLI> 13 headers, 13 lines fargo*CLI> Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK54dcd0a2;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY fargo*CLI> X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 295 v=0 o=root 20784 20786 IN IP4 10.50.103.54 s=session c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> Mar 27 11:59:29 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #27 fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK54dcd0a2;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 103 INVITE Server: Aastra 55i/2.1.0.2067 Content-Length: 0 Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 100 Trying (18) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK54dcd0a2;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 103 INVITE (16) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Content-Length: 0 (17) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:1459 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #27 - INVITE (got response) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6d577e66a409f163' Request 103: Found Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 100 to RE-invite on outgoing call 6d577e66a409f163 fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK54dcd0a2;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 103 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 316 v=0 o=MxSIP 0 31328 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK54dcd0a2;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 103 INVITE (16) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Type: application/sdp (29) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Length: 316 (19) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 31328 IN IP4 10.50.103.51 (35) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 111 10 101 (35) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:10 L16/8000 (20) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (10 headers 15 lines) --- Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 103 Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Request 103: Match Found Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6d577e66a409f163 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 10 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format L16 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x5c (ulaw|alaw|g726|slin)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:30 DEBUG[20800]: chan_sip.c:6210 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK38537cd9;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.54:1036: INVITE sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK1d67f774b8e8acd55 Max-Forwards: 70 From: ;tag=2593427753 To: "Robert Arritt" ;tag=as1d7ac186 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 26771 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Supported: timer User-Agent: Aastra 35i/20070327 Content-Type: application/sdp Content-Length: 564 v=0 o=MxSIP 0 1 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3000 RTP/AVP 0 106 107 113 110 99 10 98 97 111 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:99 PCMA/16000 a=rtpmap:10 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendonly Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42511@10.50.103.10:6050 SIP/2.0 (42) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK1d67f774b8e8acd55 (66) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) fargo*CLI> Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: ;tag=2593427753 (64) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: "Robert Arritt" ;tag=as1d7ac186 (64) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 26771 INVITE (18) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk, hold, conference (36) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Contact: Robert Arritt (66) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Supported: timer (16) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Type: application/sdp (29) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: Content-Length: 564 (19) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 14: (0) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 1 IN IP4 10.50.103.54 (31) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 106 107 113 110 99 10 98 97 111 96 9 8 101 (65) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:106 BV16/8000 (22) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:107 BV32/16000 (23) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:113 L16/16000 (22) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:110 PCMU/16000 (23) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:99 PCMA/16000 (22) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:10 L16/8000 (20) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:9 G722/8000 (20) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendonly (10) --- (14 headers 24 lines) --- Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received INVITE (5) - Command in SIP INVITE Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1015 parse_sip_options: Begin: parsing SIP "Supported: timer" Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1027 parse_sip_options: Found SIP option: -timer- Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1033 parse_sip_options: Matched SIP option: timer Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1044 parse_sip_options: * SIP extension value: 4 for call 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 Using INVITE request as basis request - 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 Sending to 10.50.103.54 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 99 Found RTP audio format 10 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 111 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3000 Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.54:3000 Found description format PCMU Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format G722 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x45c (ulaw|alaw|g726|slin|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:37 DEBUG[20800]: channel.c:2414 set_format: Set channel SIP/42511-081c5d68 to write format slin Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:10713 handle_request_invite: Got a SIP re-invite for call 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 We're at 10.50.103.10 port 17432 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK1d67f774b8e8acd55;received=10.50.103.54 From: ;tag=2593427753 To: "Robert Arritt" ;tag=as1d7ac186 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 26771 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 321 v=0 o=root 20784 20786 IN IP4 10.50.103.51 s=session c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 111 10 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:10 L16/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #28 fargo*CLI> Mar 27 11:59:37 DEBUG[20813]: rtp.c:1676 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' changed end address to 0.0.0.0:0 (format 1116) Mar 27 11:59:37 DEBUG[20813]: rtp.c:1678 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' changed end vaddress to 0.0.0.0:0 (format 1116) Mar 27 11:59:37 DEBUG[20813]: rtp.c:1680 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' was 10.50.103.54:3000/(format 92) Mar 27 11:59:37 DEBUG[20813]: rtp.c:1682 ast_rtp_bridge: Oooh, 'SIP/42512-081cc328' was 0.0.0.0:0/(format 92) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:13127 sip_set_rtp_peer: Sending reinvite on SIP '6d577e66a409f163' - It's audio soon redirected to IP 10.50.103.10 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 17328 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 (56) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK577e9317;rport (63) fargo*CLI> Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 104 INVITE (16) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 240 (19) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20787 IN IP4 10.50.103.10 (38) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.10 (21) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 17328 RTP/AVP 0 8 101 (29) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK577e9317;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 20784 20787 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 17328 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:37 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #29 fargo*CLI> Mar 27 11:59:37 DEBUG[20788]: res_musiconhold.c:557 monmp3thread: Read 104 bytes of audio while expecting 1600 fargo*CLI> <-- SIP read from 10.50.103.54:1036: ACK sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK0ce216a542c5b6652 Max-Forwards: 70 From: ;tag=2593427753 To: "Robert Arritt" ;tag=as1d7ac186 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 26771 ACK User-Agent: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: ACK sip:42511@10.50.103.10:6050 SIP/2.0 (39) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK0ce216a542c5b6652 (66) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: ;tag=2593427753 (64) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: "Robert Arritt" ;tag=as1d7ac186 (64) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 26771 ACK (15) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Length: 0 (17) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: (0) --- (9 headers 0 lines) --- Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received ACK (6) - Command in SIP ACK Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' of Response 26771: Match Found fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK577e9317;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 104 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 13704 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK577e9317;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 104 INVITE (16) fargo*CLI> Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Type: application/sdp (29) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Length: 260 (19) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 13704 IN IP4 10.50.103.51 (35) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (10 headers 13 lines) --- Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 104 Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Request 104: Match Found Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6d577e66a409f163 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:37 DEBUG[20800]: chan_sip.c:6210 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK227c821e;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> Mar 27 11:59:37 DEBUG[20813]: rtp.c:411 ast_rtcp_read: Got RTCP report of 280 bytes fargo*CLI> Mar 27 11:59:37 NOTICE[20788]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! fargo*CLI> Mar 27 11:59:37 DEBUG[20788]: res_musiconhold.c:557 monmp3thread: Read 65 bytes of audio while expecting 1600 fargo*CLI> Mar 27 11:59:37 DEBUG[20788]: res_musiconhold.c:557 monmp3thread: Read 39 bytes of audio while expecting 1600 fargo*CLI> Mar 27 11:59:38 DEBUG[20813]: channel.c:2035 ast_read: Auto-deactivating generator Mar 27 11:59:38 DEBUG[20813]: channel.c:2414 set_format: Set channel SIP/42511-081c5d68 to write format ulaw fargo*CLI> <-- SIP read from 10.50.103.54:1036: INVITE sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKf4fee92d801886584 Max-Forwards: 70 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 Call-ID: 87eec7f875b47124 CSeq: 30774 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Supported: timer User-Agent: Aastra 35i/20070327 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3002 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 (50) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKf4fee92d801886584 (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: Robert Arritt ;tag=9a4f96a8dd (72) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: 42514 (47) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 87eec7f875b47124 (25) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 30774 INVITE (18) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk, hold, conference (36) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Contact: Robert Arritt (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Supported: timer (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Type: application/sdp (29) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: Content-Length: 568 (19) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 14: (0) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 0 IN IP4 10.50.103.54 (31) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3002 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 (67) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:106 BV16/8000 (22) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:107 BV32/16000 (23) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:113 L16/16000 (22) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:110 PCMU/16000 (23) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 PCMA/16000 (23) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:112 L16/8000 (21) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:115 G726-32/8000 (25) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:9 G722/8000 (20) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (14 headers 24 lines) --- Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3191 sip_alloc: Allocating new SIP dialog for 87eec7f875b47124 - INVITE (With RTP) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received INVITE (5) - Command in SIP INVITE Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1015 parse_sip_options: Begin: parsing SIP "Supported: timer" Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1027 parse_sip_options: Found SIP option: -timer- Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1033 parse_sip_options: Matched SIP option: timer Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1044 parse_sip_options: * SIP extension value: 4 for call 87eec7f875b47124 Using INVITE request as basis request - 87eec7f875b47124 Sending to 10.50.103.54 : 5060 (non-NAT) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKf4fee92d801886584;received=10.50.103.54 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 ;tag=as057f2a6b Call-ID: 87eec7f875b47124 CSeq: 30774 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a456cb9" Content-Length: 0 --- fargo*CLI> Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #30 Scheduling destruction of call '87eec7f875b47124' in 15000 ms Found user '42512' fargo*CLI> <-- SIP read from 10.50.103.54:1036: ACK sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKf4fee92d801886584 Max-Forwards: 70 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 ;tag=as057f2a6b Call-ID: 87eec7f875b47124 CSeq: 30774 ACK User-Agent: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: ACK sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 (47) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bKf4fee92d801886584 (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: Robert Arritt ;tag=9a4f96a8dd (72) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: 42514 ;tag=as057f2a6b (62) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 87eec7f875b47124 (25) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 30774 ACK (15) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Length: 0 (17) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: (0) --- (9 headers 0 lines) --- Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received ACK (6) - Command in SIP ACK Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #30 Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '87eec7f875b47124' of Response 30774: Match Found fargo*CLI> <-- SIP read from 10.50.103.54:1036: INVITE sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK621006f98e36c3a49 Proxy-Authorization: Digest username="42512",realm="asterisk",nonce="3a456cb9",uri="sip:42514@fargo.ana.aastra.com:6050",response="ef3ccf199d6e75279c04d426353b9634",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 Call-ID: 87eec7f875b47124 CSeq: 30775 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Supported: timer User-Agent: Aastra 35i/20070327 Content-Type: application/sdp Content-Length: 568 v=0 o=MxSIP 0 0 IN IP4 10.50.103.54 s=SIP Call c=IN IP4 10.50.103.54 t=0 0 m=audio 3002 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42514@fargo.ana.aastra.com:6050 SIP/2.0 (50) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK621006f98e36c3a49 (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Proxy-Authorization: Digest username="42512",realm="asterisk",nonce="3a456cb9",uri="sip:42514@fargo.ana.aastra.com:6050",response="ef3ccf199d6e75279c04d426353b9634",algorithm=MD5 (178) fargo*CLI> Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: Max-Forwards: 70 (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: From: Robert Arritt ;tag=9a4f96a8dd (72) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: To: 42514 (47) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Call-ID: 87eec7f875b47124 (25) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: CSeq: 30775 INVITE (18) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Allow-Events: talk, hold, conference (36) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Contact: Robert Arritt (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Supported: timer (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: Content-Type: application/sdp (29) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 14: Content-Length: 568 (19) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 15: (0) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 0 IN IP4 10.50.103.54 (31) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.54 (21) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3002 RTP/AVP 0 106 107 113 110 111 112 98 97 115 96 9 8 101 (67) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:106 BV16/8000 (22) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:107 BV32/16000 (23) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:113 L16/16000 (22) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:110 PCMU/16000 (23) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:111 PCMA/16000 (23) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:112 L16/8000 (21) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:115 G726-32/8000 (25) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:9 G722/8000 (20) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (15 headers 24 lines) --- Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 87eec7f875b47124 Sending to 10.50.103.54 : 5060 (non-NAT) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Found user '42512' Found RTP audio format 0 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.54:3002 Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.54:3002 Found description format PCMU Found description format BV16 Found description format BV32 Found description format L16 Found description format PCMU Found description format PCMA Found description format L16 Found description format G726-16 Found description format G726-24 Found description format G726-32 Found description format G726-40 Found description format G722 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x45c (ulaw|alaw|g726|slin|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:10669 handle_request_invite: Checking SIP call limits for device 42512 Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:2228 update_call_counter: Updating call counter for incoming call Looking for 42514 in default (domain fargo.ana.aastra.com) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:6267 build_route: build_route: Contact hop: Robert Arritt list_route: hop: Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK621006f98e36c3a49;received=10.50.103.54 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 Call-ID: 87eec7f875b47124 CSeq: 30775 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- fargo*CLI> Mar 27 11:59:40 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 Mar 27 11:59:40 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42512 - state 2 (In use) Mar 27 11:59:40 DEBUG[20820]: app_queue.c:500 changethread: Device 'SIP/42512' changed to state '2' (In use) but we don't care because they're not a member of any queue. Mar 27 11:59:40 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: pbx.c:1697 pbx_extension_helper: Launching 'Dial' fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3191 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: channel.c:2908 ast_channel_inherit_variables: Not copying variable STACK-default-42514-1. fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPCALLID. fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPURI. fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:2083 sip_call: Outgoing Call for 42514 fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call fargo*CLI> We're at 10.50.103.10 port 12906 fargo*CLI> Adding codec 0x4 (ulaw) to SDP fargo*CLI> Adding codec 0x8 (alaw) to SDP fargo*CLI> Adding codec 0x2 (gsm) to SDP fargo*CLI> Adding non-codec 0x1 (telephone-event) to SDP fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42514@10.50.103.52;transport=udp SIP/2.0 (51) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport (63) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as7000d6c8 (66) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 3: To: (42) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 (54) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 6: CSeq: 102 INVITE (16) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 9: Date: Tue, 27 Mar 2007 15:59:40 GMT (35) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 263 (19) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3442 parse_request: Header 13: (0) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: v=0 (3) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: o=root 20784 20784 IN IP4 10.50.103.10 (38) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: s=session (9) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.10 (21) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: m=audio 12906 RTP/AVP 0 8 3 101 (31) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: a=rtpmap:3 GSM/8000 (19) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) fargo*CLI> 13 headers, 12 lines fargo*CLI> Reliably Transmitting (no NAT) to 10.50.103.52:5060: INVITE sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport From: "Robert Arritt" ;tag=as7000d6c8 To: Contact: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Mar 2007 15:59:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 20784 20784 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 12906 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> Mar 27 11:59:40 DEBUG[20819]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #32 fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 100 Trying Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as7000d6c8 To: ;tag=2c048409c43e25e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 100 Trying (18) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 (54) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 INVITE (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as7000d6c8 (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=2c048409c43e25e (48) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport (63) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1459 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #32 - INVITE (got response) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '45e36b722734fca14e4992e61d2bede0@10.50.103.10' Request 102: Found Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 100 to standard invite fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 180 Ringing Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as7000d6c8 To: ;tag=2c048409c43e25e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport Content-Length: 0 Call-Info: ;appearance-index=1 Allow-Events: talk, hold, conference Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 180 Ringing (19) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 (54) fargo*CLI> Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 INVITE (16) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as7000d6c8 (66) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=2c048409c43e25e (48) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport (63) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Call-Info: ;appearance-index=1 (56) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk, hold, conference (36) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Contact: 42514 (39) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: (0) --- (11 headers 0 lines) --- Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '45e36b722734fca14e4992e61d2bede0@10.50.103.10' Request 102: Found Mar 27 11:59:40 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 180 to standard invite Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK621006f98e36c3a49;received=10.50.103.54 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 ;tag=as099a6365 Call-ID: 87eec7f875b47124 CSeq: 30775 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Mar 27 11:59:40 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 Mar 27 11:59:40 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42514 - state 6 (Ringing) fargo*CLI> Mar 27 11:59:40 DEBUG[20821]: app_queue.c:500 changethread: Device 'SIP/42514' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Mar 27 11:59:40 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 fargo*CLI> <-- SIP read from 10.50.103.54:1036: REFER sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK5610b1c2167d3168b Max-Forwards: 70 From: ;tag=2593427753 To: "Robert Arritt" ;tag=as1d7ac186 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 26772 REFER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Robert Arritt Refer-To: 42514 Supported: timer User-Agent: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: REFER sip:42511@10.50.103.10:6050 SIP/2.0 (41) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK5610b1c2167d3168b (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: ;tag=2593427753 (64) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: "Robert Arritt" ;tag=as1d7ac186 (64) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: CSeq: 26772 REFER (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk, hold, conference (36) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Contact: Robert Arritt (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Refer-To: 42514 (53) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Supported: timer (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 14: (0) --- (14 headers 0 lines) --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received REFER (9) - Command in SIP REFER Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:10819 handle_request_refer: SIP call transfer received for call 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (REFER)! Mar 27 11:59:41 WARNING[20800]: chan_sip.c:6936 get_refer_info: Referred-by: Huh? Not a SIP header () Ignoring? Transfer to 42514 in default Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:7014 get_refer_info: Unsupervised transfer to (Refer-To): 42514 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:7017 get_refer_info: Transfer Contact Info Robert Arritt (REFER_CONTACT) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:10841 handle_request_refer: 202 Accepted (blind) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:10846 handle_request_refer: Got SIP blind transfer, applying to 'SIP/42511-081c5d68' fargo*CLI> Mar 27 11:59:41 DEBUG[20800]: channel.c:1148 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/42511-081c5d68' Mar 27 11:59:41 DEBUG[20813]: rtp.c:1722 ast_rtp_bridge: Oooh, got a hangup Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK5610b1c2167d3168b;received=10.50.103.54 From: ;tag=2593427753 To: "Robert Arritt" ;tag=as1d7ac186 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 26772 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Reliably Transmitting (no NAT) to 10.50.103.54:5060: NOTIFY sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK18aab4fb;rport From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Contact: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=26772 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #33 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.54, port 5060 Reliably Transmitting (no NAT) to 10.50.103.54:5060: BYE sip:42512@10.50.103.54:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK32294f87;rport From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #34 Mar 27 11:59:41 DEBUG[20813]: channel.c:3613 ast_channel_bridge: Returning from native bridge, channels: SIP/42511-081c5d68, SIP/42512-081cc328 Mar 27 11:59:41 DEBUG[20813]: channel.c:1371 ast_hangup: Hanging up channel 'SIP/42512-081cc328' Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:2440 sip_hangup: Hangup call SIP/42512-081cc328, SIP callid 2a0823de151ff8d37bc453412bbdd650@10.50.103.10) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:2448 sip_hangup: update_call_counter(42512) - decrement call limit counter Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:2228 update_call_counter: Updating call counter for incoming call Mar 27 11:59:41 DEBUG[20813]: app_dial.c:1644 dial_exec_full: Exiting with DIALSTATUS=ANSWER. Mar 27 11:59:41 DEBUG[20813]: pbx.c:2336 __ast_pbx_run: Spawn extension (default,42514,0) exited non-zero on 'SIP/42511-081c5d68' Mar 27 11:59:41 DEBUG[20813]: pbx.c:1697 pbx_extension_helper: Launching 'Dial' Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3191 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable STACK-default-42514-1. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable DIALSTATUS. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable DIALEDTIME. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable BLINDTRANSFER. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable BRIDGEPEER. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable STACK-default-42512-1. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPCALLID. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Mar 27 11:59:41 DEBUG[20813]: channel.c:2908 ast_channel_inherit_variables: Not copying variable SIPURI. Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:2083 sip_call: Outgoing Call for 42514 Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call We're at 10.50.103.10 port 10328 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42514@10.50.103.52;transport=udp SIP/2.0 (51) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport (63) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as3c77e5c9 (66) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: (42) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 102 INVITE (16) fargo*CLI> Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Date: Tue, 27 Mar 2007 15:59:41 GMT (35) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 263 (19) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20784 IN IP4 10.50.103.10 (38) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.10 (21) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 10328 RTP/AVP 0 8 3 101 (31) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 12 lines Reliably Transmitting (no NAT) to 10.50.103.52:5060: INVITE sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport From: "Robert Arritt" ;tag=as3c77e5c9 To: Contact: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 27 Mar 2007 15:59:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 20784 20784 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 10328 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:41 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #35 fargo*CLI> Mar 27 11:59:41 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 Mar 27 11:59:41 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42512 - state 2 (In use) Mar 27 11:59:41 DEBUG[20822]: app_queue.c:500 changethread: Device 'SIP/42512' changed to state '2' (In use) but we don't care because they're not a member of any queue. Mar 27 11:59:41 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 fargo*CLI> <-- SIP read from 10.50.103.54:1036: BYE sip:42514@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK43be473edde161c89 Proxy-Authorization: Digest username="42512",realm="asterisk",nonce="3a456cb9",uri="sip:42514@10.50.103.10:6050",response="b19eeecc353208f3787405d62f1f7861",algorithm=MD5 Max-Forwards: 70 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 ;tag=as099a6365 Call-ID: 87eec7f875b47124 CSeq: 30776 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Supported: timer User-Agent: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: BYE sip:42514@10.50.103.10:6050 SIP/2.0 (39) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK43be473edde161c89 (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Proxy-Authorization: Digest username="42512",realm="asterisk",nonce="3a456cb9",uri="sip:42514@10.50.103.10:6050",response="b19eeecc353208f3787405d62f1f7861",algorithm=MD5 (170) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: Max-Forwards: 70 (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: From: Robert Arritt ;tag=9a4f96a8dd (72) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: To: 42514 ;tag=as099a6365 (62) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Call-ID: 87eec7f875b47124 (25) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: CSeq: 30776 BYE (15) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO (88) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Allow-Events: talk, hold, conference (36) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Supported: timer (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: User-Agent: Aastra 35i/20070327 (31) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: (0) --- (13 headers 0 lines) --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 10.50.103.54 : 5060 (non-NAT) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:10958 handle_request_bye: Received bye, issuing owner hangup Transmitting (no NAT) to 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.54:5060;branch=z9hG4bK43be473edde161c89;received=10.50.103.54 From: Robert Arritt ;tag=9a4f96a8dd To: 42514 ;tag=as099a6365 Call-ID: 87eec7f875b47124 CSeq: 30776 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- fargo*CLI> Mar 27 11:59:41 DEBUG[20819]: channel.c:1371 ast_hangup: Hanging up channel 'SIP/42514-081e6e00' Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:2440 sip_hangup: Hangup call SIP/42514-081e6e00, SIP callid 45e36b722734fca14e4992e61d2bede0@10.50.103.10) Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:2448 sip_hangup: update_call_counter(42514) - decrement call limit counter Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call Scheduling destruction of call '45e36b722734fca14e4992e61d2bede0@10.50.103.10' in 32000 ms Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '45e36b722734fca14e4992e61d2bede0@10.50.103.10' of Request 102: Match Found Reliably Transmitting (no NAT) to 10.50.103.52:5060: CANCEL sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport From: "Robert Arritt" ;tag=as7000d6c8 To: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #37 Mar 27 11:59:41 DEBUG[20819]: app_dial.c:1644 dial_exec_full: Exiting with DIALSTATUS=CANCEL. Mar 27 11:59:41 DEBUG[20819]: pbx.c:2336 __ast_pbx_run: Spawn extension (default,42514,1) exited non-zero on 'SIP/42512-081e18c0' fargo*CLI> Mar 27 11:59:41 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 Mar 27 11:59:41 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42514 - state 2 (In use) Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '"Robert Arritt" <42512>' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '42512' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '42514' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'default' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/42512-081e18c0' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/42514-081e6e00' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Dial' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/42514' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-03-27 11:59:40' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-03-27 11:59:41' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '0' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'NO ANSWER' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1175011180.2' Mar 27 11:59:41 DEBUG[20819]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Mar 27 11:59:41 ERROR[20819]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied Mar 27 11:59:41 DEBUG[20819]: channel.c:1371 ast_hangup: Hanging up channel 'SIP/42512-081e18c0' Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:2440 sip_hangup: Hangup call SIP/42512-081e18c0, SIP callid 87eec7f875b47124) Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:2448 sip_hangup: update_call_counter(42512) - decrement call limit counter Mar 27 11:59:41 DEBUG[20819]: chan_sip.c:2228 update_call_counter: Updating call counter for incoming call Mar 27 11:59:41 DEBUG[20823]: app_queue.c:500 changethread: Device 'SIP/42514' changed to state '2' (In use) but we don't care because they're not a member of any queue. Mar 27 11:59:41 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 Mar 27 11:59:41 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 Mar 27 11:59:41 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42512 - state 1 (Not in use) Mar 27 11:59:41 DEBUG[20824]: app_queue.c:500 changethread: Device 'SIP/42512' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Mar 27 11:59:41 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42512 fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 400 Missing status-line Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK18aab4fb;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 104 NOTIFY Server: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 400 Missing status-line (31) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK18aab4fb;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 104 NOTIFY (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Server: Aastra 35i/20070327 (27) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- fargo*CLI> Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' of Request 104: Match Found Destroying call '87eec7f875b47124' fargo*CLI> <-- SIP read from 10.50.103.54:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK32294f87;rport=6050;received=10.50.103.10 From: "Robert Arritt" ;tag=as1d7ac186 To: ;tag=2593427753 Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 CSeq: 105 BYE Server: Aastra 35i/20070327 Content-Length: 0 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK32294f87;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as1d7ac186 (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=2593427753 (62) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 2a0823de151ff8d37bc453412bbdd650@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 105 BYE (13) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Server: Aastra 35i/20070327 (27) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' of Request 105: Match Found Destroying call '2a0823de151ff8d37bc453412bbdd650@10.50.103.10' fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 200 OK Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 CANCEL From: "Robert Arritt" ;tag=as7000d6c8 To: ;tag=2c048409c43e25e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport Content-Length: 0 Contact: 42514 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 CANCEL (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as7000d6c8 (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=2c048409c43e25e (48) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport (63) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Contact: 42514 (39) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Supported: replaces (19) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) --- (10 headers 0 lines) --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #37 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '45e36b722734fca14e4992e61d2bede0@10.50.103.10' of Request 102: Match Found fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 487 Request Terminated Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as7000d6c8 To: ;tag=2c048409c43e25e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 INVITE (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as7000d6c8 (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=2c048409c43e25e (48) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport (63) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- fargo*CLI> Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '45e36b722734fca14e4992e61d2bede0@10.50.103.10' of Request 102: Match Not Found Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 10.50.103.52:5060: ACK sip:42514@10.50.103.52;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK3f2551b2;rport From: "Robert Arritt" ;tag=as7000d6c8 To: ;tag=2c048409c43e25e Contact: Call-ID: 45e36b722734fca14e4992e61d2bede0@10.50.103.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 100 Trying Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 100 Trying (18) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 INVITE (16) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as3c77e5c9 (66) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=a4224231a06449e (48) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport (63) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1459 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #35 - INVITE (got response) Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' Request 102: Found Mar 27 11:59:41 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 100 to standard invite fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 180 Ringing Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport Content-Length: 0 Call-Info: ;appearance-index=1 Allow-Events: talk, hold, conference Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 180 Ringing (19) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 INVITE (16) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as3c77e5c9 (66) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=a4224231a06449e (48) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport (63) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Call-Info: ;appearance-index=1 (56) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk, hold, conference (36) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Contact: 42514 (39) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: (0) --- (11 headers 0 lines) --- Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' Request 102: Found Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 180 to standard invite fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: channel.c:2087 ast_indicate: Driver for channel 'SIP/42511-081c5d68' does not support indication 3, emulating it fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: channel.c:2414 set_format: Set channel SIP/42511-081c5d68 to write format slin fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42514 - state 6 (Ringing) fargo*CLI> Mar 27 11:59:42 DEBUG[20825]: app_queue.c:500 changethread: Device 'SIP/42514' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: rtp.c:1269 ast_rtp_raw_write: Difference is 102976, ms is 12892 fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: rtp.c:411 ast_rtcp_read: Got RTCP report of 224 bytes fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 200 OK Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 102 INVITE From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport Content-Length: 241 Call-Info: ;appearance-index=1 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 v=0 o=MxSIP 0 391943521 IN IP4 10.50.103.52 s=SIP Call c=IN IP4 10.50.103.52 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 102 INVITE (16) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as3c77e5c9 (66) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=a4224231a06449e (48) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK0d18f4af;rport (63) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 241 (19) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Call-Info: ;appearance-index=1 (56) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow-Events: talk,hold,conference (34) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Content-Type: application/sdp (29) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Supported: replaces (19) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: Contact: 42514 (39) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 14: (0) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 391943521 IN IP4 10.50.103.52 (39) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.52 (21) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) fargo*CLI> --- (14 headers 11 lines) --- fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' of Request 102: Match Found fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 200 to standard invite fargo*CLI> Found RTP audio format 0 fargo*CLI> Found RTP audio format 8 fargo*CLI> Found RTP audio format 101 fargo*CLI> Peer audio RTP is at port 10.50.103.52:3000 fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.52:3000 fargo*CLI> Found description format PCMU fargo*CLI> Found description format PCMA fargo*CLI> Found description format telephone-event fargo*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) fargo*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) fargo*CLI> Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:6267 build_route: build_route: Contact hop: 42514 fargo*CLI> list_route: hop: fargo*CLI> set_destination: Parsing for address/port to send to fargo*CLI> set_destination: set destination to 10.50.103.52, port 5060 fargo*CLI> Transmitting (no NAT) to 10.50.103.52:5060: ACK sip:42514@10.50.103.52 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK7343cb94;rport From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Contact: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: channel.c:2414 set_format: Set channel SIP/42511-081c5d68 to write format ulaw fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/42514-081ec340' fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:13127 sip_set_rtp_peer: Sending reinvite on SIP '6d577e66a409f163' - It's audio soon redirected to IP 10.50.103.52 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 17328 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 (56) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK61a9a176;rport (63) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 105 INVITE (16) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 239 (19) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20788 IN IP4 10.50.103.52 (38) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.52 (21) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK61a9a176;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 105 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 20784 20788 IN IP4 10.50.103.52 s=session c=IN IP4 10.50.103.52 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #38 Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:13127 sip_set_rtp_peer: Sending reinvite on SIP '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' - It's audio soon redirected to IP 10.50.103.51 set_destination: Parsing for address/port to send to fargo*CLI> set_destination: set destination to 10.50.103.52, port 5060 fargo*CLI> We're at 10.50.103.10 port 10328 fargo*CLI> Adding codec 0x4 (ulaw) to SDP fargo*CLI> Adding codec 0x8 (alaw) to SDP fargo*CLI> Adding non-codec 0x1 (telephone-event) to SDP fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42514@10.50.103.52 SIP/2.0 (37) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK1be9ceca;rport (63) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: "Robert Arritt" ;tag=as3c77e5c9 (66) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: ;tag=a4224231a06449e (62) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 103 INVITE (16) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 239 (19) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20785 IN IP4 10.50.103.51 (38) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) fargo*CLI> 13 headers, 11 lines fargo*CLI> Reliably Transmitting (no NAT) to 10.50.103.52:5060: INVITE sip:42514@10.50.103.52 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK1be9ceca;rport From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Contact: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 20784 20785 IN IP4 10.50.103.51 s=session c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #39 fargo*CLI> Mar 27 11:59:42 DEBUG[20813]: rtp.c:1361 ast_rtp_write: Ooh, format changed from unknown to ulaw fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/42514-081ec340' fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42514 - state 2 (In use) fargo*CLI> Mar 27 11:59:42 DEBUG[20826]: app_queue.c:500 changethread: Device 'SIP/42514' changed to state '2' (In use) but we don't care because they're not a member of any queue. fargo*CLI> Mar 27 11:59:42 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 100 Trying Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 103 INVITE From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK1be9ceca;rport Content-Length: 0 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 100 Trying (18) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 103 INVITE (16) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as3c77e5c9 (66) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=a4224231a06449e (48) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK1be9ceca;rport (63) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 0 (17) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:1459 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #39 - INVITE (got response) Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' Request 103: Found Mar 27 11:59:42 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 100 to RE-invite on outgoing call 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK61a9a176;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 105 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 16806 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK61a9a176;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 105 INVITE (16) fargo*CLI> Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Type: application/sdp (29) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Length: 260 (19) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 16806 IN IP4 10.50.103.51 (35) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (10 headers 13 lines) --- Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 105 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Request 105: Match Found Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6d577e66a409f163 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:6210 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 fargo*CLI> Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK5bd6968f;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: SIP/2.0 200 OK Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 103 INVITE From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK1be9ceca;rport Content-Length: 241 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: 42514 User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 v=0 o=MxSIP 0 391943521 IN IP4 10.50.103.52 s=SIP Call c=IN IP4 10.50.103.52 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: CSeq: 103 INVITE (16) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: From: "Robert Arritt" ;tag=as3c77e5c9 (66) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: ;tag=a4224231a06449e (48) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK1be9ceca;rport (63) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Content-Length: 241 (19) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Allow-Events: talk,hold,conference (34) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Type: application/sdp (29) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Supported: replaces (19) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: Contact: 42514 (39) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) fargo*CLI> Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 391943521 IN IP4 10.50.103.52 (39) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.52 (21) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 11 lines) --- Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 103 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' of Request 103: Match Found Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:9718 handle_response_invite: SIP response 200 to RE-invite on outgoing call 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.52:3000 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.52:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:6210 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.52, port 5060 Transmitting (no NAT) to 10.50.103.52:5060: ACK sip:42514@10.50.103.52 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4c1037b5;rport From: "Robert Arritt" ;tag=as3c77e5c9 To: ;tag=a4224231a06449e Contact: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- fargo*CLI> <-- SIP read from 10.50.103.52:5060: BYE sip:42511@10.50.103.10:6050 SIP/2.0 Via: SIP/2.0/UDP 10.50.103.52;branch=z9hG4bK31b3838b0 Max-Forwards: 70 Content-Length: 0 To: "Robert Arritt" ;tag=as3c77e5c9 From: ;tag=a4224231a06449e Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 1096396721 BYE Supported: timer Call-Info: ;appearance-index=1 Supported: replaces User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: BYE sip:42511@10.50.103.10:6050 SIP/2.0 (39) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.52;branch=z9hG4bK31b3838b0 (53) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: Max-Forwards: 70 (16) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: Content-Length: 0 (17) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: To: "Robert Arritt" ;tag=as3c77e5c9 (64) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: From: ;tag=a4224231a06449e (50) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 (54) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: CSeq: 1096396721 BYE (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Supported: timer (16) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Call-Info: ;appearance-index=1 (56) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: Supported: replaces (19) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 11: User-Agent: Aastra 480i Cordless/1.4.2.1073 Brcm Callctrl/1.5 MxSF/v3.2.8.45 (76) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 12: (0) --- (12 headers 0 lines) --- Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:11318 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 10.50.103.52 : 5060 (non-NAT) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:10958 handle_request_bye: Received bye, issuing owner hangup Transmitting (no NAT) to 10.50.103.52:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.52;branch=z9hG4bK31b3838b0;received=10.50.103.52 fargo*CLI> From: ;tag=a4224231a06449e To: "Robert Arritt" ;tag=as3c77e5c9 Call-ID: 4bf25a3a464732b770ce06672de91cb7@10.50.103.10 CSeq: 1096396721 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: rtp.c:1722 ast_rtp_bridge: Oooh, got a hangup Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:13127 sip_set_rtp_peer: Sending reinvite on SIP '6d577e66a409f163' - It's audio soon redirected to IP 10.50.103.10 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 We're at 10.50.103.10 port 17328 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 (56) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6a9efd75;rport (63) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 4: Contact: (38) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 5: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 6: CSeq: 106 INVITE (16) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 7: User-Agent: Asterisk PBX (24) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 8: Max-Forwards: 70 (16) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 11: Content-Type: application/sdp (29) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 12: Content-Length: 240 (19) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3442 parse_request: Header 13: (0) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: o=root 20784 20789 IN IP4 10.50.103.10 (38) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: s=session (9) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.10 (21) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: m=audio 17328 RTP/AVP 0 8 101 (29) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-16 (15) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.50.103.51:5060: INVITE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6a9efd75;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 106 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 20784 20789 IN IP4 10.50.103.10 s=session c=IN IP4 10.50.103.10 t=0 0 m=audio 17328 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #40 Mar 27 11:59:43 DEBUG[20813]: channel.c:3613 ast_channel_bridge: Returning from native bridge, channels: SIP/42511-081c5d68, SIP/42514-081ec340 Mar 27 11:59:43 DEBUG[20813]: channel.c:1371 ast_hangup: Hanging up channel 'SIP/42514-081ec340' Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:2440 sip_hangup: Hangup call SIP/42514-081ec340, SIP callid 4bf25a3a464732b770ce06672de91cb7@10.50.103.10) Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:2448 sip_hangup: update_call_counter(42514) - decrement call limit counter Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call Mar 27 11:59:43 DEBUG[20813]: app_dial.c:1644 dial_exec_full: Exiting with DIALSTATUS=ANSWER. Mar 27 11:59:43 DEBUG[20813]: pbx.c:2336 __ast_pbx_run: Spawn extension (default,42514,1) exited non-zero on 'SIP/42511-081c5d68' Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '"Robert Arritt" <42511>' Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '42511' Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '42514' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'default' fargo*CLI> Mar 27 11:59:43 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 Mar 27 11:59:43 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42514 - state 1 (Not in use) Mar 27 11:59:43 DEBUG[20827]: app_queue.c:500 changethread: Device 'SIP/42514' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Mar 27 11:59:43 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42514 fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/42511-081c5d68' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/42514-081ec340' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Dial' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/42514' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-03-27 11:59:27' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-03-27 11:59:28' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-03-27 11:59:43' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '16' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '15' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1175011167.0' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' fargo*CLI> Mar 27 11:59:43 ERROR[20813]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: channel.c:1371 ast_hangup: Hanging up channel 'SIP/42511-081c5d68' fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:2440 sip_hangup: Hangup call SIP/42511-081c5d68, SIP callid 6d577e66a409f163) fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:2448 sip_hangup: update_call_counter(42511) - decrement call limit counter fargo*CLI> Mar 27 11:59:43 DEBUG[20813]: chan_sip.c:2228 update_call_counter: Updating call counter for outgoing call fargo*CLI> Scheduling destruction of call '6d577e66a409f163' in 32000 ms fargo*CLI> Mar 27 11:59:43 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42511 Mar 27 11:59:43 DEBUG[20787]: devicestate.c:187 do_state_change: Changing state for SIP/42511 - state 1 (Not in use) Mar 27 11:59:43 DEBUG[20828]: app_queue.c:500 changethread: Device 'SIP/42511' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Mar 27 11:59:43 DEBUG[20787]: chan_sip.c:11866 sip_devicestate: Checking device state for peer 42511 fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6a9efd75;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 106 INVITE Contact: Robert Arritt Server: Aastra 55i/2.1.0.2067 Content-Type: application/sdp Content-Length: 260 v=0 o=MxSIP 0 14437 IN IP4 10.50.103.51 s=SIP Call c=IN IP4 10.50.103.51 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK6a9efd75;rport=6050;received=10.50.103.10 (90) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 106 INVITE (16) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Contact: Robert Arritt (66) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: Content-Type: application/sdp (29) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 9: Content-Length: 260 (19) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 10: (0) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: v=0 (3) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: o=MxSIP 0 14437 IN IP4 10.50.103.51 (35) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: s=SIP Call (10) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: c=IN IP4 10.50.103.51 (21) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: t=0 0 (5) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=silenceSupp:on - - - - (24) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=fmtp:101 0-15 (15) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=ptime:30 (10) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3474 parse_request: Line: a=sendrecv (10) --- (10 headers 13 lines) --- Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1392 __sip_ack: Acked pending invite 106 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Request 106: Match Found Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:9720 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.103.51:3000 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3684 process_sdp: Peer audio RTP is at port 10.50.103.51:3000 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:6210 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Transmitting (no NAT) to 10.50.103.51:5060: ACK sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK4904743c;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Contact: Call-ID: 6d577e66a409f163 CSeq: 106 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.103.51, port 5060 Reliably Transmitting (no NAT) to 10.50.103.51:5060: BYE sip:42511@10.50.103.51:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK5e7d1a0d;rport From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 107 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1307 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #42 Scheduling destruction of call '6d577e66a409f163' in 32000 ms Destroying call '4bf25a3a464732b770ce06672de91cb7@10.50.103.10' fargo*CLI> <-- SIP read from 10.50.103.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK5e7d1a0d;rport=6050;received=10.50.103.10 From: 42512 ;tag=as24d4e1f6 To: Robert Arritt ;tag=f44e6a847d Call-ID: 6d577e66a409f163 CSeq: 107 BYE Server: Aastra 55i/2.1.0.2067 Content-Length: 0 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 0: SIP/2.0 200 OK (14) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.103.10:6050;branch=z9hG4bK5e7d1a0d;rport=6050;received=10.50.103.10 (90) fargo*CLI> Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 2: From: 42512 ;tag=as24d4e1f6 (64) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 3: To: Robert Arritt ;tag=f44e6a847d (70) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 4: Call-ID: 6d577e66a409f163 (25) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 5: CSeq: 107 BYE (13) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 6: Server: Aastra 55i/2.1.0.2067 (29) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 7: Content-Length: 0 (17) Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:3442 parse_request: Header 8: (0) --- (8 headers 0 lines) --- Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1403 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 Mar 27 11:59:43 DEBUG[20800]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '6d577e66a409f163' of Request 107: Match Found Destroying call '6d577e66a409f163' fargo*CLI>