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- [Jun 17 16:09:17] VERBOSE[9636] chan_sip.c: Really destroying SIP dialog '1146327453@xxx.xxx.xxx.xxx' Method: OPTIONS
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://208.94.157.10:5060 --->
- INVITE sip:17168980077@xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 1 INVITE
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39
- Max-Forwards: 68
- P-Asserted-Identity: <sip:7169074915@cxc.dashcs.com:5060>
- Supported: 100rel
- Content-Disposition: session;handling=optional
- Contact: <sip:7169074915@208.94.157.10:5060;transport=udp>
- Min-SE: 900
- Session-Expires: 1800
- Content-Type: application/sdp
- Content-Length: 238
- v=0
- o=Acme_UAS 0 1 IN IP4 208.94.157.10
- s=SIP Media Capabilities
- c=IN IP4 208.94.157.10
- t=0 0
- m=audio 22126 RTP/AVP 0 18 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:18 G729/8000
- a=rtpmap:101 telephone-event/8000
- a=maxptime:20
- a=sendrecv
- <------------->
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: --- (15 headers 11 lines) ---
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Sending to 208.94.157.10 : 5060 (NAT)
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Using INVITE request as basis request - CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found peer '4-208.94.157.10' for '7169074915' from 208.94.157.10:5060
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found RTP audio format 0
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found RTP audio format 18
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found RTP audio format 101
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 208.94.157.10:22126
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found audio description format PCMU for ID 0
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found audio description format G729 for ID 18
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found audio description format telephone-event for ID 101
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Capabilities: us - 0xec6 (gsm|ulaw|g726|slin|lpc10|speex|ilbc), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 208.94.157.10:22126
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Looking for 17168980077 in from-external (domain xxx.xxx.xxx.xxx)
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: list_route: hop: <sip:7169074915@208.94.157.10:5060;transport=udp>
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c:
- <--- Transmitting (no NAT) to 208.94.157.10:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39;received=208.94.157.10
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 1 INVITE
- Server: i-Communicate
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:17168980077@xxx.xxx.xxx.xxx>
- Content-Length: 0
- <------------>
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Audio is at xxx.xxx.xxx.xxx port 16414
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x1000 (g722) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x8 (alaw) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x40 (slin) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x2 (gsm) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x800 (g726) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x20 (adpcm) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x80 (lpc10) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
- INVITE sip:01163956@xxx.xxx.xxx.xxx SIP/2.0
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b2cd08d;rport
- Max-Forwards: 70
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: <sip:01163956@xxx.xxx.xxx.xxx>
- Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- CSeq: 102 INVITE
- User-Agent: i-Communicate
- Date: Thu, 17 Jun 2010 20:09:19 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Diversion: <sip:7168980077@betapbx.i-evolve.com>
- Content-Type: application/sdp
- Content-Length: 406
- v=0
- o=root 801668698 801668698 IN IP4 xxx.xxx.xxx.xxx
- s=Asterisk PBX 1.6.1.6
- c=IN IP4 xxx.xxx.xxx.xxx
- t=0 0
- m=audio 16414 RTP/AVP 0 9 8 10 3 111 5 7 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:9 G722/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:10 L16/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:111 G726-32/8000
- a=rtpmap:5 DVI4/8000
- a=rtpmap:7 LPC/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- SIP/2.0 100 Giving a try
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b2cd08d;rport=5060
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: <sip:01163956@xxx.xxx.xxx.xxx>
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- CSeq: 102 INVITE
- Server: Enswitch SIP proxy
- Content-Length: 0
- Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=7889 req_src_ip=xxx.xxx.xxx.xxx req_src_port=5060 in_uri=sip:01163956@xxx.xxx.xxx.xxx out_uri=sip:01163956@72.237.213.162:5060 via_cnt==1"
- <------------->
- [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: --- (9 headers 0 lines) ---
- [Jun 17 16:09:19] VERBOSE[6650] chan_sip.c:
- <--- Transmitting (no NAT) to 208.94.157.10:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39;received=208.94.157.10
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 1 INVITE
- Server: i-Communicate
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:17168980077@xxx.xxx.xxx.xxx>
- Content-Length: 0
- <------------>
- [Jun 17 16:09:20] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK0b2cd08d;rport=5060
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- CSeq: 102 INVITE
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- Contact: <sip:01163956@72.237.213.162:5060>
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
- Allow-Events: talk,hold,conference
- Accept-Language: en
- Content-Length: 0
- <------------->
- [Jun 17 16:09:20] VERBOSE[9636] chan_sip.c: --- (12 headers 0 lines) ---
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK0b2cd08d;rport=5060
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- CSeq: 102 INVITE
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- Contact: <sip:01163956@72.237.213.162:5060>
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
- Supported: 100rel,replaces
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
- Accept-Language: en
- Content-Type: application/sdp
- Content-Length: 223
- v=0
- o=- 1276804589 1276804589 IN IP4 72.237.213.162
- s=Polycom IP Phone
- c=IN IP4 72.237.213.162
- t=0 0
- m=audio 2224 RTP/AVP 0 127
- a=sendrecv
- a=rtpmap:0 PCMU/8000
- a=rtpmap:127 telephone-event/8000
- a=direction:active
- <------------->
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: --- (14 headers 10 lines) ---
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found RTP audio format 0
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found RTP audio format 127
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found audio description format PCMU for ID 0
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found audio description format telephone-event for ID 127
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Capabilities: us - 0x1eee (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|ilbc|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: list_route: hop: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: set_destination: Parsing <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748> for address/port to send to
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: set_destination: set destination to xxx.xxx.xxx.xxx, port 5060
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
- ACK sip:01163956@72.237.213.162:5060 SIP/2.0
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK1b713980;rport
- Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- Max-Forwards: 70
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- CSeq: 102 ACK
- User-Agent: i-Communicate
- Content-Length: 0
- ---
- [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c: Audio is at xxx.xxx.xxx.xxx port 15146
- [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c:
- <--- Reliably Transmitting (no NAT) to 208.94.157.10:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39;received=208.94.157.10
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 1 INVITE
- Server: i-Communicate
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:17168980077@xxx.xxx.xxx.xxx>
- Content-Type: application/sdp
- Content-Length: 234
- v=0
- o=root 1777021475 1777021475 IN IP4 xxx.xxx.xxx.xxx
- s=Asterisk PBX 1.6.1.6
- c=IN IP4 xxx.xxx.xxx.xxx
- t=0 0
- m=audio 15146 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------>
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://208.94.157.10:5060 --->
- ACK sip:17168980077@xxx.xxx.xxx.xxx SIP/2.0
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 1 ACK
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f26e-4c1a80f0-2435ab2a-5efdfff
- Max-Forwards: 69
- Contact: <sip:7169074915@208.94.157.10:5060;transport=udp>
- Content-Length: 0
- <------------->
- [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: --- (9 headers 0 lines) ---
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- ACK sip:7169074915@xxx.xxx.xxx.xxx SIP/2.0
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK316d00ef28E9F3DA
- Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bK316d00ef28E9F3DA
- From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- CSeq: 1 ACK
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- Contact: <sip:01163956@72.237.213.162:5060>
- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
- Accept-Language: en
- Max-Forwards: 69
- Content-Length: 0
- X-Enswitch-RURI: sip:7169074915@xxx.xxx.xxx.xxx
- X-Enswitch-Source: 72.237.213.162:5060
- <------------->
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: --- (17 headers 0 lines) ---
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- INVITE sip:7169074915@xxx.xxx.xxx.xxx SIP/2.0
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.0
- Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bKcd850595F1DA3120
- From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- CSeq: 2 INVITE
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- Contact: <sip:01163956@72.237.213.162:5060>
- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
- Accept-Language: en
- Supported: 100rel,replaces
- Allow-Events: talk,hold,conference
- Proxy-Authorization: Digest username="01163956", realm="xxx.xxx.xxx.xxx", nonce="4c1a81100000aab2b38bba0eb354d50ebda8f8920e08a560", uri="sip:7169074915@xxx.xxx.xxx.xxx", response="eeba5533d886d1d79c9d25983b1ad637", algorithm=MD5
- Max-Forwards: 69
- Content-Type: application/sdp
- Content-Length: 215
- X-Enswitch-RURI: sip:7169074915@xxx.xxx.xxx.xxx
- X-Enswitch-Source: 72.237.213.162:5060
- v=0
- o=- 1276804589 1276804590 IN IP4 72.237.213.162
- s=Polycom IP Phone
- c=IN IP4 72.237.213.162
- t=0 0
- a=sendonly
- m=audio 2224 RTP/AVP 0 101
- a=sendonly
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- <------------->
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: --- (21 headers 10 lines) ---
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found RTP audio format 0
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found RTP audio format 101
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found audio description format PCMU for ID 0
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found audio description format telephone-event for ID 101
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Capabilities: us - 0x1eee (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|ilbc|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
- <--- Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.0;received=xxx.xxx.xxx.xxx
- Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bKcd850595F1DA3120
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
- From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- CSeq: 2 INVITE
- Server: i-Communicate
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
- Content-Length: 0
- <------------>
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Audio is at xxx.xxx.xxx.xxx port 16414
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
- <--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.0;received=xxx.xxx.xxx.xxx
- Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bKcd850595F1DA3120
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
- From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- CSeq: 2 INVITE
- Server: i-Communicate
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
- Content-Type: application/sdp
- Content-Length: 232
- v=0
- o=root 801668698 801668699 IN IP4 xxx.xxx.xxx.xxx
- s=Asterisk PBX 1.6.1.6
- c=IN IP4 xxx.xxx.xxx.xxx
- t=0 0
- m=audio 16414 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=recvonly
- <------------>
- [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- ACK sip:7169074915@xxx.xxx.xxx.xxx SIP/2.0
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.2
- Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bK35274ca15EB5ED6C
- From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- CSeq: 2 ACK
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- Contact: <sip:01163956@72.237.213.162:5060>
- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
- Accept-Language: en
- Max-Forwards: 69
- Content-Length: 0
- X-Enswitch-RURI: sip:7169074915@xxx.xxx.xxx.xxx
- X-Enswitch-Source: 72.237.213.162:5060
- <--- SIP read from UDP://208.94.157.10:5060 --->
- BYE sip:17168980077@xxx.xxx.xxx.xxx SIP/2.0
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 2 BYE
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f334-4c1a80fa-2435d106-bde9e0e
- Max-Forwards: 69
- Content-Length: 0
- <------------->
- [Jun 17 16:09:31] VERBOSE[9636] chan_sip.c: --- (8 headers 0 lines) ---
- [Jun 17 16:09:31] VERBOSE[9636] chan_sip.c: Sending to 208.94.157.10 : 5060 (no NAT)
- [Jun 17 16:09:31] VERBOSE[9636] chan_sip.c:
- <--- Transmitting (no NAT) to 208.94.157.10:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f334-4c1a80fa-2435d106-bde9e0e;received=208.94.157.10
- From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
- To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
- Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
- CSeq: 2 BYE
- Server: i-Communicate
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: Scheduling destruction of SIP dialog '496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx' in 32000 ms (Method: ACK)
- [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: set_destination: Parsing <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748> for address/port to send to
- [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: set_destination: set destination to xxx.xxx.xxx.xxx, port 5060
- [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
- BYE sip:01163956@72.237.213.162:5060 SIP/2.0
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK07359977;rport
- Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- Max-Forwards: 70
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- CSeq: 103 BYE
- User-Agent: i-Communicate
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- [Jun 17 16:09:32] VERBOSE[9636] chan_sip.c:
- <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK07359977;rport=5060
- From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
- To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
- CSeq: 103 BYE
- Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
- Contact: <sip:01163956@72.237.213.162:5060>
- Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
- Accept-Language: en
- Content-Length: 0
- <------------>
- [Jun 17 16:10:05] VERBOSE[9636] chan_sip.c: Scheduling destruction of SIP dialog '547214668@xxx.xxx.xxx.xxx' in 32000 ms (Method: OPTIONS)
- [Jun 17 16:10:07] VERBOSE[9636] chan_sip.c: Really destroying SIP dialog '757803964@xxx.xxx.xxx.xxx' Method: OPTIONS
- [Jun 17 16:10:15] VERBOSE[9636] chan_sip.c:
- <------------->
- [Jun 17 16:10:15] VERBOSE[9636] chan_sip.c: --- (11 headers 0 lines) ---
- [Jun 17 16:10:15] VERBOSE[9636] chan_sip.c: Looking for s in from-external (domain xxx.xxx.xxx.xxx)
- [Jun 17 16:10:15] VERBOSE[9636] chan_sip.c:
- <------------>
- [Jun 17 16:10:15] VERBOSE[9636] chan_sip.c: Scheduling destruction of SIP dialog '1769911092@xxx.xxx.xxx.xxx' in 32000 ms (Method: OPTIONS)
- [Jun 17 16:10:17] VERBOSE[9636] chan_sip.c: Really destroying SIP dialog '349366721@xxx.xxx.xxx.xxx' Method: OPTIONS
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