[Jul 6 13:45:31] VERBOSE[7968] config.c: == Parsing '/etc/asterisk/logger.conf': [Jul 6 13:45:31] VERBOSE[7968] config.c: == Found [Jul 6 13:45:31] VERBOSE[7968] logger.c: Asterisk Event Logger restarted [Jul 6 13:45:31] VERBOSE[7968] logger.c: Asterisk Queue Logger restarted [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.18:5060 ---> INVITE sip:42@111.111.111.221 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.18:5060;rport;branch=z9hG4bK31635 From: ;tag=13321 To: Call-ID: 25497 CSeq: 20 INVITE Contact: Content-Type: application/sdp Max-Forwards: 70 User-Agent: Linphone/3.1.0 (eXosip2/3.3.0) Subject: Phone call Content-Length: 399 v=0 o=From1011 123456 654321 IN IP4 111.111.111.18 s=A conversation c=IN IP4 111.111.111.18 t=0 0 m=audio 7078 RTP/AVP 111 110 0 3 8 101 a=rtpmap:111 speex/16000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 98 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 <-------------> [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: --- (12 headers 16 lines) --- [Jul 6 13:45:35] VERBOSE[7958] netsock.c: == Using SIP RTP CoS mark 5 [Jul 6 13:45:35] VERBOSE[7958] netsock.c: == Using SIP VRTP CoS mark 6 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Sending to 111.111.111.18 : 5060 (NAT) [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Using INVITE request as basis request - 25497 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found peer 'From1011' for 'From1011' from 111.111.111.18:5060 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP audio format 111 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP audio format 110 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP audio format 0 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP audio format 3 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP audio format 8 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP audio format 101 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found RTP video format 98 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Peer audio RTP is at port 111.111.111.18:7078 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found audio description format speex for ID 111 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found audio description format speex for ID 110 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found audio description format GSM for ID 3 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found audio description format PCMA for ID 8 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Found video description format H263-1998 for ID 98 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Capabilities: us - 0xc7f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x100000 (h263p)/text=0x0 (nothing), combined - 0x10020e (gsm|ulaw|alaw|speex|h263p) [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Peer audio RTP is at port 111.111.111.18:7078 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Peer video RTP is at port 111.111.111.18:9078 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: Looking for 42 in from-sip (domain 111.111.111.221) [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: list_route: hop: [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: <--- Transmitting (no NAT) to 111.111.111.18:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.111.111.18:5060;branch=z9hG4bK31635;received=111.111.111.18;rport=5060 From: ;tag=13321 To: Call-ID: 25497 CSeq: 20 INVITE Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 6 13:45:35] VERBOSE[7975] pbx.c: -- Executing [42@from-sip:1] Dial("SIP/From1011-b0062298", "SIP/Station42") in new stack [Jul 6 13:45:35] VERBOSE[7975] netsock.c: == Using SIP RTP CoS mark 5 [Jul 6 13:45:35] VERBOSE[7975] netsock.c: == Using SIP VRTP CoS mark 6 [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Audio is at 111.111.111.221 port 16406 [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Video is at 111.111.111.221 port 14646 [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: We think we can do text [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x10 (g726aal2) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x20 (adpcm) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x40 (slin) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x80 (lpc10) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x800 (g726) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x1000 (g722) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding codec 0x8000 (slin16) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: Reliably Transmitting (no NAT) to 111.111.111.205:5060: INVITE sip:Station42@111.111.111.205 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK358cab6a;rport Max-Forwards: 70 From: "From 1011 Account" ;tag=as3e3688a4 To: Contact: Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.1 Date: Mon, 06 Jul 2009 11:45:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 556 v=0 o=root 1987274863 1987274863 IN IP4 111.111.111.221 s=Asterisk PBX 1.6.1.1 c=IN IP4 111.111.111.221 b=CT:384 t=0 0 m=audio 16406 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 14646 RTP/AVP 98 a=rtpmap:98 h263-1998/90000 a=sendrecv --- [Jul 6 13:45:35] VERBOSE[7975] app_dial.c: -- Called Station42 [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.205:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK358cab6a;rport From: "From 1011 Account" ;tag=as3e3688a4 To: ;tag=3958E7C2-F57158D7 CSeq: 102 INVITE Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 Contact: User-Agent: PolycomVVX-VVX_1500-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 <-------------> [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.205:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK358cab6a;rport From: "From 1011 Account" ;tag=as3e3688a4 To: ;tag=3958E7C2-F57158D7 CSeq: 102 INVITE Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 Contact: User-Agent: PolycomVVX-VVX_1500-UA/3.1.3.0439 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 <-------------> [Jul 6 13:45:35] VERBOSE[7958] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 13:45:35] VERBOSE[7975] app_dial.c: -- SIP/Station42-0e738a88 is ringing [Jul 6 13:45:35] VERBOSE[7975] chan_sip.c: <--- Transmitting (no NAT) to 111.111.111.18:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.111.111.18:5060;branch=z9hG4bK31635;received=111.111.111.18;rport=5060 From: ;tag=13321 To: ;tag=as7fc3432b Call-ID: 25497 CSeq: 20 INVITE Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK358cab6a;rport From: "From 1011 Account" ;tag=as3e3688a4 To: ;tag=3958E7C2-F57158D7 CSeq: 102 INVITE Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_1500-UA/3.1.3.0439 Accept-Language: en Content-Type: application/sdp Content-Length: 278 v=0 o=- 1246884173 1246884173 IN IP4 111.111.111.205 s=Polycom IP Phone c=IN IP4 111.111.111.205 b=AS:384 t=0 0 m=audio 2332 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 98 a=sendrecv a=rtpmap:98 h263-1998/90000 <-------------> [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: --- (12 headers 13 lines) --- [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Found RTP audio format 0 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Found RTP audio format 101 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Found RTP video format 98 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Peer audio RTP is at port 111.111.111.205:2332 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Found video description format h263-1998 for ID 98 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Capabilities: us - 0xc7f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140), peer - audio=0x4 (ulaw)/video=0x100000 (h263p)/text=0x0 (nothing), combined - 0x100004 (ulaw|h263p) [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Peer audio RTP is at port 111.111.111.205:2332 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: list_route: hop: [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: set_destination: set destination to 111.111.111.205, port 5060 [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: Transmitting (no NAT) to 111.111.111.205:5060: ACK sip:Station42@111.111.111.205 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK581fef64;rport Max-Forwards: 70 From: "From 1011 Account" ;tag=as3e3688a4 To: ;tag=3958E7C2-F57158D7 Contact: Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.1 Content-Length: 0 --- [Jul 6 13:45:36] VERBOSE[7975] app_dial.c: -- SIP/Station42-0e738a88 answered SIP/From1011-b0062298 [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Audio is at 111.111.111.221 port 14006 [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Video is at 111.111.111.221 port 11580 [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Adding codec 0x200 (speex) to SDP [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 13:45:36] VERBOSE[7975] chan_sip.c: <--- Reliably Transmitting (no NAT) to 111.111.111.18:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.111.18:5060;branch=z9hG4bK31635;received=111.111.111.18;rport=5060 From: ;tag=13321 To: ;tag=as7fc3432b Call-ID: 25497 CSeq: 20 INVITE Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 422 v=0 o=root 1074158595 1074158595 IN IP4 111.111.111.221 s=Asterisk PBX 1.6.1.1 c=IN IP4 111.111.111.221 b=CT:384 t=0 0 m=audio 14006 RTP/AVP 3 0 8 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11580 RTP/AVP 98 a=rtpmap:98 h263-1998/90000 a=sendrecv <------------> [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.18:5060 ---> ACK sip:42@111.111.111.221 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.18:5060;rport;branch=z9hG4bK16502 From: ;tag=13321 To: ;tag=as7fc3432b Call-ID: 25497 CSeq: 20 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.1.0 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jul 6 13:45:36] VERBOSE[7958] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.205:5060 ---> BYE sip:016@111.111.111.221 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.205;branch=z9hG4bKfaf2b6cF3DE6EB9 From: ;tag=3958E7C2-F57158D7 To: "From 1011 Account" ;tag=as3e3688a4 CSeq: 1 BYE Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 Contact: User-Agent: PolycomVVX-VVX_1500-UA/3.1.3.0439 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: Sending to 111.111.111.205 : 5060 (no NAT) [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: <--- Transmitting (no NAT) to 111.111.111.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.111.205;branch=z9hG4bKfaf2b6cF3DE6EB9;received=111.111.111.205 From: ;tag=3958E7C2-F57158D7 To: "From 1011 Account" ;tag=as3e3688a4 Call-ID: 5514681638999f0a1d7bf1521acdca8a@111.111.111.221 CSeq: 1 BYE Server: Asterisk PBX 1.6.1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> [Jul 6 13:45:43] VERBOSE[7975] pbx.c: == Spawn extension (from-sip, 42, 1) exited non-zero on 'SIP/From1011-b0062298' [Jul 6 13:45:43] VERBOSE[7975] chan_sip.c: Scheduling destruction of SIP dialog '25497' in 7488 ms (Method: ACK) [Jul 6 13:45:43] VERBOSE[7975] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 13:45:43] VERBOSE[7975] chan_sip.c: set_destination: set destination to 111.111.111.18, port 5060 [Jul 6 13:45:43] VERBOSE[7975] chan_sip.c: Reliably Transmitting (no NAT) to 111.111.111.18:5060: BYE sip:From1011@111.111.111.18:5060 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK27c63b5a;rport Max-Forwards: 70 From: ;tag=as7fc3432b To: ;tag=13321 Call-ID: 25497 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: <--- SIP read from UDP://111.111.111.18:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.111.221:5060;branch=z9hG4bK27c63b5a;rport=5060 From: ;tag=as7fc3432b To: ;tag=13321 Call-ID: 25497 CSeq: 102 BYE User-Agent: Linphone/3.1.0 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: --- (8 headers 0 lines) --- [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: Really destroying SIP dialog '25497' Method: ACK [Jul 6 13:45:43] VERBOSE[7958] chan_sip.c: Really destroying SIP dialog '5514681638999f0a1d7bf1521acdca8a@111.111.111.221' Method: BYE [Jul 6 13:53:55] VERBOSE[7958] chan_sip.c: -- Unregistered SIP 'From1011'