[Mar 21 17:23:44] Asterisk SVN-branch-12-r410933 built by root @ newtonr-laptop on a x86_64 running Linux on 2014-03-19 19:33:12 UTC [Mar 21 17:23:44] VERBOSE[10291] config.c: == Parsing '/etc/asterisk/logger.conf': Found [Mar 21 17:23:44] VERBOSE[10291] logger.c: Asterisk Queue Logger restarted [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjf6ouUHIpfS7Qnz3ea96WMSRhzNap27FC ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH ˙To: "RustyONE" ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35642 REGISTER ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyONE" ˙Expires: 300 ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjf6ouUHIpfS7Qnz3ea96WMSRhzNap27FC;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH ˙To: "RustyONE" ;tag=as7501571f ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35642 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ea88942" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'm6qybWzN.pQsFuItAEreYfCYc7g34OiV' in 32000 ms (Method: REGISTER) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjp6QWZY61f-cZAtFc6QZnXXHoRW08emya ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH ˙To: "RustyONE" ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35643 REGISTER ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyONE" ˙Expires: 300 ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Authorization: Digest username="6001", realm="asterisk", nonce="0ea88942", uri="sip:10.24.18.124:5060", response="0d9a4cf998fecb18a1dc1aeada67936d", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: > Saved useragent "Digium D40 1_4_0_0_57389" for peer 6001 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjp6QWZY61f-cZAtFc6QZnXXHoRW08emya;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH ˙To: "RustyONE" ;tag=as7501571f ˙Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV ˙CSeq: 35643 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 300 ˙Contact: ;expires=300 ˙Date: Fri, 21 Mar 2014 22:23:53 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'm6qybWzN.pQsFuItAEreYfCYc7g34OiV' in 32000 ms (Method: REGISTER) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjNfwx-c7QwuGq84MDyveExyrXTac9WtHF ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW ˙To: "RustyONE" ˙Contact: "RustyONE" ˙Call-ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU ˙CSeq: 28240 SUBSCRIBE ˙Event: message-summary ˙Expires: 3600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/simple-message-summary ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjNfwx-c7QwuGq84MDyveExyrXTac9WtHF;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW ˙To: "RustyONE" ;tag=as077662f6 ˙Call-ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU ˙CSeq: 28240 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01cd83e3" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'EfpQTvraASBU5bJNxdvadNsbie4hQPdU' in 32000 ms (Method: SUBSCRIBE) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjrsgb9myTX73MzE5HMv.anTfgLpzDm37V ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H ˙To: sip:6001@10.24.18.124 ˙Contact: "RustyONE" ˙Call-ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh ˙CSeq: 13230 SUBSCRIBE ˙Event: presence ˙Expires: 600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/pidf+xml, application/xpidf+xml ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjrsgb9myTX73MzE5HMv.anTfgLpzDm37V;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H ˙To: sip:6001@10.24.18.124;tag=as71af05d0 ˙Call-ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh ˙CSeq: 13230 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b448484" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog 'ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh' in 32000 ms (Method: SUBSCRIBE) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjinQ0qrKMjxH9EkGyIwsrfPukKgeuGue3 ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW ˙To: "RustyONE" ˙Contact: "RustyONE" ˙Call-ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU ˙CSeq: 28241 SUBSCRIBE ˙Event: message-summary ˙Expires: 3600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/simple-message-summary ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Authorization: Digest username="6001", realm="asterisk", nonce="01cd83e3", uri="sip:6001@10.24.18.124:5060", response="700a90c1b9948867dbcfe4d9266ea1fd", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 404 Not found (no mailbox) ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPjinQ0qrKMjxH9EkGyIwsrfPukKgeuGue3;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW ˙To: "RustyONE" ;tag=as077662f6 ˙Call-ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU ˙CSeq: 28241 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:53] NOTICE[10274] chan_sip.c: Received SIP subscribe for peer without mailbox: 6001 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'EfpQTvraASBU5bJNxdvadNsbie4hQPdU' Method: SUBSCRIBE [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPje49HN9pC1NWMm2Iknk31s1xzDwE0gqU4 ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H ˙To: sip:6001@10.24.18.124 ˙Contact: "RustyONE" ˙Call-ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh ˙CSeq: 13231 SUBSCRIBE ˙Event: presence ˙Expires: 600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: application/pidf+xml, application/xpidf+xml ˙Allow-Events: presence, message-summary, refer ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Authorization: Digest username="6001", realm="asterisk", nonce="5b448484", uri="sip:6001@10.24.18.124:5060", response="ac53da48636015c11b92d0ce0b5e8a2b", algorithm=MD5 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Looking for 6001 in from-internal (domain 10.24.18.124) [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 404 Not Found ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPje49HN9pC1NWMm2Iknk31s1xzDwE0gqU4;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H ˙To: sip:6001@10.24.18.124;tag=as71af05d0 ˙Call-ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh ˙CSeq: 13231 SUBSCRIBE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh' Method: SUBSCRIBE [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙INVITE sip:6002@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj76WseEzga1EaojalP5Cij5JMboutQq7Z ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ˙Contact: "RustyONE" ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20917 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Type: application/sdp ˙Content-Length: 288 ˙ ˙v=0 ˙o=- 90811929 90811929 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4020 RTP/AVP 0 8 9 96 ˙a=rtcp:4021 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (15 headers 14 lines) --- [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Using INVITE request as basis request - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj76WseEzga1EaojalP5Cij5JMboutQq7Z;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ;tag=as7333f32b ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20917 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="208379fb" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' in 32000 ms (Method: INVITE) [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙ACK sip:6002@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj76WseEzga1EaojalP5Cij5JMboutQq7Z ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ;tag=as7333f32b ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20917 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙INVITE sip:6002@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj-8x1t4VTHr0yC-A26sPV4zQbT5GWbIDi ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ˙Contact: "RustyONE" ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20918 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Authorization: Digest username="6001", realm="asterisk", nonce="208379fb", uri="sip:6002@10.24.18.124", response="005d7cb521c26e6ee7ee44d797ac65ac", algorithm=MD5 ˙Content-Type: application/sdp ˙Content-Length: 288 ˙ ˙v=0 ˙o=- 90811929 90811929 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4020 RTP/AVP 0 8 9 96 ˙a=rtcp:4021 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Using INVITE request as basis request - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found peer '6001' for '6001' from 10.24.18.16:5060 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 8 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 9 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format G722 for ID 9 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4020 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Looking for 6002 in from-internal (domain 10.24.18.124) [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj-8x1t4VTHr0yC-A26sPV4zQbT5GWbIDi;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20918 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:57] VERBOSE[10294][C-00000000] pbx.c: -- Executing [6002@from-internal:1] Dial("SIP/6001-00000000", "SIP/6002,15") in new stack [Mar 21 17:23:57] VERBOSE[10294][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:23:57] VERBOSE[10294][C-00000000] chan_sip.c: Audio is at 15552 [Mar 21 17:23:57] VERBOSE[10294][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:23:57] VERBOSE[10294][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:23:57] VERBOSE[10294][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c404372 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Date: Fri, 21 Mar 2014 22:23:57 GMT ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1111986011 1111986011 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 15552 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:23:57] VERBOSE[10294][C-00000000] app_dial.c: -- Called SIP/6002 [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK6c404372 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ˙CSeq: 102 INVITE ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 180 ringing ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK6c404372 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙CSeq: 102 INVITE ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:57] VERBOSE[10294][C-00000000] app_dial.c: -- SIP/6002-00000001 is ringing [Mar 21 17:23:57] VERBOSE[10294][C-00000000] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 180 Ringing ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj-8x1t4VTHr0yC-A26sPV4zQbT5GWbIDi;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ;tag=as28bb3d0e ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20918 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:23:59] VERBOSE[10294][C-00000000] res_rtp_asterisk.c: > 0x7f5dc800fa40 -- Probation passed - setting RTP source address to 10.24.18.138:4010 [Mar 21 17:23:59] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK6c404372 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙CSeq: 102 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Contact: "RustyTWO" ˙Supported: replaces, 100rel, timer, norefersub ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90811929 90811930 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4010 RTP/AVP 0 96 ˙a=rtcp:4011 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:23:59] VERBOSE[10274] chan_sip.c: --- (13 headers 12 lines) --- [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4010 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK02e7b545 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:23:59] VERBOSE[10294][C-00000000] app_dial.c: -- SIP/6002-00000001 answered SIP/6001-00000000 [Mar 21 17:23:59] VERBOSE[10294][C-00000000] chan_sip.c: Audio is at 18866 [Mar 21 17:23:59] VERBOSE[10294][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:23:59] VERBOSE[10294][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:23:59] VERBOSE[10294][C-00000000] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.16:5060;branch=z9hG4bKPj-8x1t4VTHr0yC-A26sPV4zQbT5GWbIDi;received=10.24.18.16;rport=5060 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ;tag=as28bb3d0e ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20918 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Type: application/sdp ˙Require: timer ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1133247888 1133247888 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 18866 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙<------------> [Mar 21 17:23:59] VERBOSE[10294][C-00000000] bridge_channel.c: -- Channel SIP/6001-00000000 joined 'simple_bridge' basic-bridge <48397e75-23cd-41a2-9a69-6d056da1192b> [Mar 21 17:23:59] VERBOSE[10295][C-00000000] bridge_channel.c: -- Channel SIP/6002-00000001 joined 'simple_bridge' basic-bridge <48397e75-23cd-41a2-9a69-6d056da1192b> [Mar 21 17:23:59] VERBOSE[10295][C-00000000] bridge.c: > Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: switching from simple_bridge technology to native_rtp [Mar 21 17:23:59] VERBOSE[10295][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:23:59] VERBOSE[10295][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:23:59] VERBOSE[10295][C-00000000] chan_sip.c: Audio is at 15552 [Mar 21 17:23:59] VERBOSE[10295][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:23:59] VERBOSE[10295][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:23:59] VERBOSE[10295][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK45fe1d4e ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1111986011 1111986012 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:23:59] VERBOSE[10295][C-00000000] res_rtp_asterisk.c: > 0x7f5dc800fa40 -- Probation passed - setting RTP source address to 10.24.18.138:4010 [Mar 21 17:24:00] VERBOSE[10294][C-00000000] res_rtp_asterisk.c: > 0x7f5d68030030 -- Probation passed - setting RTP source address to 10.24.18.16:4020 [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙ACK sip:6002@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjvbR6GkgAYHFpM0imQ4GRtQV4C8bhmKgH ˙Max-Forwards: 70 ˙From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙To: ;tag=as28bb3d0e ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 20918 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Audio is at 18866 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK494166cd;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Contact: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 289 ˙ ˙v=0 ˙o=root 1133247888 1133247889 IN IP4 10.24.18.138 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙m=audio 4010 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK45fe1d4e ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙CSeq: 103 INVITE ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90811929 90811931 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4010 RTP/AVP 0 96 ˙a=rtcp:4011 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4010 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3bf8775b ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' Method: REGISTER [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK494166cd ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙CSeq: 102 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyONE" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 240 ˙ ˙v=0 ˙o=- 90811929 90811930 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtcp:4021 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4020 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ˙ACK sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK48a58160;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Contact: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:00] VERBOSE[10295][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:00] VERBOSE[10295][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:00] VERBOSE[10295][C-00000000] chan_sip.c: Audio is at 15552 [Mar 21 17:24:00] VERBOSE[10295][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:00] VERBOSE[10295][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:00] VERBOSE[10295][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54805a0f ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 104 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1111986011 1111986013 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK54805a0f ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙CSeq: 104 INVITE ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90811929 90811932 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4010 RTP/AVP 0 96 ˙a=rtcp:4011 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4010 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK38749d38 ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 104 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙INVITE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjKaKBJBea7CyKmCFJZ5EuVktXMAeYGyMt ˙Max-Forwards: 70 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Contact: "RustyTWO" ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14655 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Type: application/sdp ˙Content-Length: 303 ˙ ˙v=0 ˙o=- 90811929 90811933 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙a=sendonly ˙m=audio 4010 RTP/AVP 0 8 9 96 ˙a=rtcp:4011 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙a=sendonly ˙<-------------> [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: --- (15 headers 15 lines) --- [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 8 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 9 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format G722 for ID 9 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4010 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjKaKBJBea7CyKmCFJZ5EuVktXMAeYGyMt;received=10.24.18.138;rport=5060 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14655 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Audio is at 15552 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjKaKBJBea7CyKmCFJZ5EuVktXMAeYGyMt;received=10.24.18.138;rport=5060 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14655 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Contact: ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1111986011 1111986014 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=recvonly ˙ ˙<------------> [Mar 21 17:24:07] VERBOSE[10294][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:07] VERBOSE[10294][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:07] VERBOSE[10294][C-00000000] chan_sip.c: Audio is at 18866 [Mar 21 17:24:07] VERBOSE[10294][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:07] VERBOSE[10294][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:07] VERBOSE[10294][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK66296f91;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Contact: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1133247888 1133247890 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 18866 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:07] VERBOSE[10294][C-00000000] res_musiconhold.c: -- Started music on hold, class 'default', on channel 'SIP/6001-00000000' [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙ACK sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjwRlymbhpLreWlq3LISobkELSLLRRe-lZ ˙Max-Forwards: 70 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14655 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:07] VERBOSE[10294][C-00000000] res_rtp_asterisk.c: > 0x7f5d68030030 -- Probation passed - setting RTP source address to 10.24.18.16:4020 [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK66296f91 ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙CSeq: 103 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyONE" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 240 ˙ ˙v=0 ˙o=- 90811929 90811931 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtcp:4021 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4020 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ˙ACK sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK547c07fa;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Contact: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Audio is at 18866 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK05a16f19;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Contact: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 104 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1133247888 1133247891 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 18866 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:07] VERBOSE[10294][C-00000000] res_rtp_asterisk.c: > 0x7f5d68030030 -- Probation passed - setting RTP source address to 10.24.18.16:4020 [Mar 21 17:24:08] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK05a16f19 ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙CSeq: 104 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyONE" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 240 ˙ ˙v=0 ˙o=- 90811929 90811932 IN IP4 10.24.18.16 ˙s=digphn ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtcp:4021 IN IP4 10.24.18.16 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:08] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Peer audio RTP is at port 10.24.18.16:4020 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Transmitting (no NAT) to 10.24.18.16:5060: ˙ACK sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK1435713e;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Contact: ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 104 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:08] VERBOSE[10294][C-00000000] res_rtp_asterisk.c: > 0x7f5d68030030 -- Probation passed - setting RTP source address to 10.24.18.16:4020 [Mar 21 17:24:08] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL' Method: REGISTER [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bK5fd638b9FCFDB300 ˙From: "6003" ;tag=416D771C-70EB3F17 ˙To: ˙CSeq: 1 REGISTER ˙Call-ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 ˙Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Max-Forwards: 70 ˙Expires: 300 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.180:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bK5fd638b9FCFDB300;received=10.24.18.180 ˙From: "6003" ;tag=416D771C-70EB3F17 ˙To: ;tag=as58185e37 ˙Call-ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 ˙CSeq: 1 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30c935b5" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '1096126b-1194f96a-fd46e4ad@10.24.18.180' in 32000 ms (Method: REGISTER) [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙REGISTER sip:10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bK4653224C3B24BFF ˙From: "6003" ;tag=416D771C-70EB3F17 ˙To: ˙CSeq: 2 REGISTER ˙Call-ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 ˙Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Authorization: Digest username="6003", realm="asterisk", nonce="30c935b5", uri="sip:10.24.18.124:5060", response="7bc4c5112001ed8978a8f017501a13bd", algorithm=MD5 ˙Max-Forwards: 70 ˙Expires: 300 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: > Saved useragent "PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477" for peer 6003 [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bK4653224C3B24BFF;received=10.24.18.180 ˙From: "6003" ;tag=416D771C-70EB3F17 ˙To: ;tag=as58185e37 ˙Call-ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 ˙CSeq: 2 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 300 ˙Contact: ;expires=300 ˙Date: Fri, 21 Mar 2014 22:24:09 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '1096126b-1194f96a-fd46e4ad@10.24.18.180' in 32000 ms (Method: REGISTER) [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙INVITE sip:6003@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjqdYZ6JhR5.byBIcxJhzkK68-39SU.nf6 ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ˙Contact: "RustyTWO" ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16872 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙X-Digium-Call-Hint: potentialTransfer ˙Content-Type: application/sdp ˙Content-Length: 291 ˙ ˙v=0 ˙o=- 90811942 90811942 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4012 RTP/AVP 0 8 9 96 ˙a=rtcp:4013 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Using INVITE request as basis request - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found peer '6002' for '6002' from 10.24.18.138:5060 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjqdYZ6JhR5.byBIcxJhzkK68-39SU.nf6;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ;tag=as0f822371 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16872 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="208aa590" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' in 32000 ms (Method: INVITE) [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙ACK sip:6003@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjqdYZ6JhR5.byBIcxJhzkK68-39SU.nf6 ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ;tag=as0f822371 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16872 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙INVITE sip:6003@10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPj1UIVnvufifyEK6zN5Fl2reID84W3wb54 ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ˙Contact: "RustyTWO" ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16873 INVITE ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Session-Expires: 1800 ˙Min-SE: 90 ˙User-Agent: Digium D40 1_4_0_0_57389 ˙X-Digium-Call-Hint: potentialTransfer ˙Authorization: Digest username="6002", realm="asterisk", nonce="208aa590", uri="sip:6003@10.24.18.124", response="be1414879cdfc964aac84a1d0096b286", algorithm=MD5 ˙Content-Type: application/sdp ˙Content-Length: 291 ˙ ˙v=0 ˙o=- 90811942 90811942 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4012 RTP/AVP 0 8 9 96 ˙a=rtcp:4013 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:8 PCMA/8000 ˙a=rtpmap:9 G722/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (17 headers 14 lines) --- [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Using INVITE request as basis request - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found peer '6002' for '6002' from 10.24.18.138:5060 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 8 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 9 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format G722 for ID 9 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4012 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Looking for 6003 in from-internal (domain 10.24.18.124) [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: list_route: route/path hop: [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPj1UIVnvufifyEK6zN5Fl2reID84W3wb54;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16873 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:10] VERBOSE[10296][C-00000001] pbx.c: -- Executing [6003@from-internal:1] Dial("SIP/6002-00000002", "SIP/6003,15") in new stack [Mar 21 17:24:10] VERBOSE[10296][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:24:10] VERBOSE[10296][C-00000001] chan_sip.c: Audio is at 19470 [Mar 21 17:24:10] VERBOSE[10296][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:10] VERBOSE[10296][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:10] VERBOSE[10296][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Date: Fri, 21 Mar 2014 22:24:10 GMT ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1559736078 1559736078 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 19470 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:10] VERBOSE[10296][C-00000001] app_dial.c: -- Called SIP/6003 [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 100 Trying ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 102 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 180 Ringing ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 102 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Allow-Events: talk,hold,conference ˙Accept-Language: en ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (11 headers 0 lines) --- [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: list_route: route/path hop: [Mar 21 17:24:10] VERBOSE[10296][C-00000001] app_dial.c: -- SIP/6003-00000003 is ringing [Mar 21 17:24:10] VERBOSE[10296][C-00000001] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 180 Ringing ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPj1UIVnvufifyEK6zN5Fl2reID84W3wb54;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ;tag=as5a51f621 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16873 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:12] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 102 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440649 1395440649 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2224 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:24:12] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2224 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: list_route: route/path hop: [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK578cb7bd ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:12] VERBOSE[10296][C-00000001] app_dial.c: -- SIP/6003-00000003 answered SIP/6002-00000002 [Mar 21 17:24:12] VERBOSE[10296][C-00000001] chan_sip.c: Audio is at 19846 [Mar 21 17:24:12] VERBOSE[10296][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:12] VERBOSE[10296][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:12] VERBOSE[10296][C-00000001] chan_sip.c: ˙<--- Reliably Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPj1UIVnvufifyEK6zN5Fl2reID84W3wb54;received=10.24.18.138;rport=5060 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ;tag=as5a51f621 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16873 INVITE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Session-Expires: 1800;refresher=uas ˙Contact: ˙Content-Type: application/sdp ˙Require: timer ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1876385532 1876385532 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 19846 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙<------------> [Mar 21 17:24:12] VERBOSE[10296][C-00000001] bridge_channel.c: -- Channel SIP/6002-00000002 joined 'simple_bridge' basic-bridge [Mar 21 17:24:12] VERBOSE[10297][C-00000001] bridge_channel.c: -- Channel SIP/6003-00000003 joined 'simple_bridge' basic-bridge [Mar 21 17:24:12] VERBOSE[10297][C-00000001] bridge.c: > Bridge e501657b-704b-4b2f-9d8f-55487218673b: switching from simple_bridge technology to native_rtp [Mar 21 17:24:12] VERBOSE[10297][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:12] VERBOSE[10297][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:12] VERBOSE[10297][C-00000001] chan_sip.c: Audio is at 19470 [Mar 21 17:24:12] VERBOSE[10297][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:12] VERBOSE[10297][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:12] VERBOSE[10297][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK24774eda ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 289 ˙ ˙v=0 ˙o=root 1559736078 1559736079 IN IP4 10.24.18.138 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙m=audio 4012 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:12] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK24774eda ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 103 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440649 1395440650 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2224 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:24:12] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2224 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK324ea04e ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:13] VERBOSE[10297][C-00000001] res_rtp_asterisk.c: > 0x7f5dd000fa90 -- Probation passed - setting RTP source address to 10.24.18.180:2224 [Mar 21 17:24:13] VERBOSE[10296][C-00000001] res_rtp_asterisk.c: > 0x7f5d6803ffa0 -- Probation passed - setting RTP source address to 10.24.18.138:4012 [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙ACK sip:6003@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjx8DdIatSNQEBTLVPUdcOBRO0WQZWAd.. ˙Max-Forwards: 70 ˙From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙To: ;tag=as5a51f621 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 16873 ACK ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Audio is at 19846 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3b5f4909;rport ˙Max-Forwards: 70 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙Contact: ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 102 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 289 ˙ ˙v=0 ˙o=root 1876385532 1876385533 IN IP4 10.24.18.180 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2224 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK3b5f4909 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙CSeq: 102 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90811942 90811943 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4012 RTP/AVP 0 96 ˙a=rtcp:4013 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4012 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK4650bfb9;rport ˙Max-Forwards: 70 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙Contact: ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 102 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:13] VERBOSE[10297][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:13] VERBOSE[10297][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:13] VERBOSE[10297][C-00000001] chan_sip.c: Audio is at 19470 [Mar 21 17:24:13] VERBOSE[10297][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:13] VERBOSE[10297][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:13] VERBOSE[10297][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK732a9bac ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 104 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 289 ˙ ˙v=0 ˙o=root 1559736078 1559736080 IN IP4 10.24.18.138 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙m=audio 4012 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK732a9bac ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 104 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440649 1395440651 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2224 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2224 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK517f3638 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 104 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙REFER sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjMVSpfmLL-nFUrBvMn3ZLpvDTuQbdP39j ˙Max-Forwards: 70 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Contact: "RustyTWO" ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14656 REFER ˙Event: refer ˙Expires: 600 ˙Supported: replaces, 100rel, timer, norefersub ˙Accept: message/sipfrag;version=2.0 ˙Allow-Events: presence, message-summary, refer ˙Refer-To: ˙Referred-By: ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (17 headers 0 lines) --- [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Call 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 got a SIP call transfer from caller: (REFER)! [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: SIP transfer to extension 6003@from-internal by 6002@10.24.18.138 [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 202 Accepted ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPjMVSpfmLL-nFUrBvMn3ZLpvDTuQbdP39j;received=10.24.18.138;rport=5060 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14656 REFER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Contact: ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:15] VERBOSE[10274][C-00000000] bridge_channel.c: -- Channel SIP/6001-00000000 left 'native_rtp' basic-bridge <48397e75-23cd-41a2-9a69-6d056da1192b> [Mar 21 17:24:15] VERBOSE[10274][C-00000000] bridge_channel.c: -- Channel SIP/6001-00000000 swapped with SIP/6002-00000002 into 'native_rtp' basic-bridge [Mar 21 17:24:15] VERBOSE[10274][C-00000000] bridge_channel.c: -- Channel SIP/6002-00000002 left 'native_rtp' basic-bridge [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Audio is at 19846 [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3c619fd5;rport ˙Max-Forwards: 70 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙Contact: ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 103 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Session-Expires: 1800;refresher=uac ˙Min-SE: 90 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1876385532 1876385534 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 19846 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10274][C-00000000] bridge.c: > Bridge e501657b-704b-4b2f-9d8f-55487218673b: switching from native_rtp technology to simple_bridge [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Audio is at 19470 [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3b85ed75 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 105 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 290 ˙ ˙v=0 ˙o=root 1559736078 1559736081 IN IP4 10.24.18.124 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.124 ˙t=0 0 ˙m=audio 19470 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10274][C-00000000] bridge.c: > Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: switching from native_rtp technology to simple_bridge [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙NOTIFY sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK64a01688;rport ˙Max-Forwards: 70 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙Contact: ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 105 NOTIFY ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Event: refer;id=14656 ˙Subscription-state: terminated;reason=noresource ˙Content-Type: message/sipfrag;version=2.0 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 16 ˙ ˙SIP/2.0 200 OK ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10294][C-00000000] res_musiconhold.c: -- Stopped music on hold on SIP/6001-00000000 [Mar 21 17:24:15] VERBOSE[10296][C-00000001] pbx.c: == Spawn extension (from-internal, 6003, 1) exited non-zero on 'SIP/6002-00000002' [Mar 21 17:24:15] VERBOSE[10296][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' in 32000 ms (Method: ACK) [Mar 21 17:24:15] VERBOSE[10295][C-00000000] bridge_channel.c: -- Channel SIP/6002-00000001 left 'simple_bridge' basic-bridge <48397e75-23cd-41a2-9a69-6d056da1192b> [Mar 21 17:24:15] VERBOSE[10295][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' in 32000 ms (Method: REFER) [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3b85ed75 ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 105 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440649 1395440652 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2224 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2224 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK110c0a9d ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 105 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Audio is at 19470 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.180:5060: ˙INVITE sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK47888bb4 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 106 INVITE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙X-asterisk-Info: SIP re-invite (External RTP bridge) ˙Content-Type: application/sdp ˙Content-Length: 287 ˙ ˙v=0 ˙o=root 1559736078 1559736082 IN IP4 10.24.18.16 ˙s=Asterisk PBX SVN-branch-12-r410933 ˙c=IN IP4 10.24.18.16 ˙t=0 0 ˙m=audio 4020 RTP/AVP 0 96 ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-16 ˙a=silenceSupp:off - - - - ˙a=ptime:20 ˙a=maxptime:150 ˙a=sendrecv ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10297][C-00000001] res_rtp_asterisk.c: > 0x7f5dd000fa90 -- Probation passed - setting RTP source address to 10.24.18.180:2224 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK47888bb4 ˙From: "Bob" ;tag=as56dbf74c ˙To: "6003" ;tag=884188E1-2198B888 ˙CSeq: 106 INVITE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER ˙Supported: 100rel,replaces ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Content-Type: application/sdp ˙Content-Length: 197 ˙ ˙v=0 ˙o=- 1395440649 1395440653 IN IP4 10.24.18.180 ˙s=Polycom IP Phone ˙c=IN IP4 10.24.18.180 ˙t=0 0 ˙m=audio 2224 RTP/AVP 0 96 ˙a=sendrecv ˙a=rtpmap:0 PCMU/8000 ˙a=rtpmap:96 telephone-event/8000 ˙<-------------> [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (13 headers 9 lines) --- [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.180:2224 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.180:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.180:5060: ˙ACK sip:6003@10.24.18.180:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK03930119 ˙Max-Forwards: 70 ˙From: "Bob" ;tag=as56dbf74c ˙To: ;tag=884188E1-2198B888 ˙Contact: ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 106 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10297][C-00000001] res_rtp_asterisk.c: > 0x7f5dd000fa90 -- Probation passed - setting RTP source address to 10.24.18.180:2224 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK3c619fd5 ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙CSeq: 103 INVITE ˙Session-Expires: 1800;refresher=uac ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Type: application/sdp ˙Content-Length: 243 ˙ ˙v=0 ˙o=- 90811942 90811944 IN IP4 10.24.18.138 ˙s=digphn ˙c=IN IP4 10.24.18.138 ˙t=0 0 ˙a=X-nat:0 ˙m=audio 4012 RTP/AVP 0 96 ˙a=rtcp:4013 IN IP4 10.24.18.138 ˙a=rtpmap:0 PCMU/8000 ˙a=sendrecv ˙a=rtpmap:96 telephone-event/8000 ˙a=fmtp:96 0-15 ˙<-------------> [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 96 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Peer audio RTP is at port 10.24.18.138:4012 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.24.18.138:5060: ˙ACK sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK591f26c9;rport ˙Max-Forwards: 70 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙Contact: ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 103 ACK ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: set_destination: set destination to 10.24.18.138:5060 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.138:5060: ˙BYE sip:6002@10.24.18.138:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK2fd19a3b;rport ˙Max-Forwards: 70 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙CSeq: 104 BYE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Proxy-Authorization: Digest username="6002", realm="asterisk", algorithm=MD5, uri="sip:10.24.18.124", nonce="208aa590", response="0543904cc25015ab98e67276763976be" ˙X-Asterisk-HangupCause: Normal Clearing ˙X-Asterisk-HangupCauseCode: 16 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' in 32000 ms (Method: ACK) [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK64a01688 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙From: "Alice" ;tag=as1ecd7421 ˙To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙CSeq: 105 NOTIFY ˙Contact: "RustyTWO" ˙Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ˙Supported: replaces, 100rel, timer, norefersub ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK2fd19a3b ˙Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 ˙From: ;tag=as5a51f621 ˙To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx ˙CSeq: 104 BYE ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' Method: ACK [Mar 21 17:24:16] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.138:5060 ---> ˙BYE sip:6001@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPj8aeIXHcjCpJV9ZRdJB1TPzG8p3PwMmmU ˙Max-Forwards: 70 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14657 BYE ˙User-Agent: Digium D40 1_4_0_0_57389 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:16] VERBOSE[10274] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 17:24:16] VERBOSE[10274][C-00000000] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:24:16] VERBOSE[10274][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' in 32000 ms (Method: BYE) [Mar 21 17:24:16] VERBOSE[10274][C-00000000] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.138:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.138:5060;branch=z9hG4bKPj8aeIXHcjCpJV9ZRdJB1TPzG8p3PwMmmU;received=10.24.18.138;rport=5060 ˙From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV ˙To: "Alice" ;tag=as1ecd7421 ˙Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 ˙CSeq: 14657 BYE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.166:5060 ---> ˙REGISTER sip:10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK15431221533272715572;rport ˙From: 6004 ;tag=3167523177 ˙To: 6004 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 3 REGISTER ˙Contact: ˙Max-Forwards: 70 ˙Expires: 60 ˙Supported: path ˙User-Agent: Voip Phone 1.0 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.166:5060 ---> ˙SIP/2.0 401 Unauthorized ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK15431221533272715572;received=10.24.18.166;rport=5060 ˙From: 6004 ;tag=3167523177 ˙To: 6004 ;tag=as571a4f15 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 3 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="333d9384" ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '21608363712124-1345167932089@10.24.18.166' in 32000 ms (Method: REGISTER) [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'm6qybWzN.pQsFuItAEreYfCYc7g34OiV' Method: REGISTER [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.166:5060 ---> ˙REGISTER sip:10.24.18.124 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK478224601275052328;rport ˙From: 6004 ;tag=3167523177 ˙To: 6004 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 4 REGISTER ˙Contact: ˙Authorization: Digest username="6004", realm="asterisk", nonce="333d9384", uri="sip:10.24.18.124", response="a3a03b3588cb9a622daee27cf9fac1bb", algorithm=MD5 ˙Max-Forwards: 70 ˙Expires: 60 ˙Supported: path ˙User-Agent: Voip Phone 1.0 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.166:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK478224601275052328;received=10.24.18.166;rport=5060 ˙From: 6004 ;tag=3167523177 ˙To: 6004 ;tag=as571a4f15 ˙Call-ID: 21608363712124-1345167932089@10.24.18.166 ˙CSeq: 4 REGISTER ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Expires: 60 ˙Contact: ;expires=60 ˙Date: Fri, 21 Mar 2014 22:24:25 GMT ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Scheduling destruction of SIP dialog '21608363712124-1345167932089@10.24.18.166' in 32000 ms (Method: REGISTER) [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.180:5060 ---> ˙BYE sip:6002@10.24.18.124:5060 SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bKc5b09fd3835AF732 ˙From: "6003" ;tag=884188E1-2198B888 ˙To: "Bob" ;tag=as56dbf74c ˙CSeq: 1 BYE ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙Contact: ˙User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 ˙Accept-Language: en ˙Max-Forwards: 70 ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: --- (11 headers 0 lines) --- [Mar 21 17:24:40] VERBOSE[10274][C-00000001] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:40] VERBOSE[10274][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' in 32000 ms (Method: BYE) [Mar 21 17:24:40] VERBOSE[10274][C-00000001] chan_sip.c: ˙<--- Transmitting (no NAT) to 10.24.18.180:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bKc5b09fd3835AF732;received=10.24.18.180 ˙From: "6003" ;tag=884188E1-2198B888 ˙To: "Bob" ;tag=as56dbf74c ˙Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 ˙CSeq: 1 BYE ˙Server: Asterisk PBX SVN-branch-12-r410933 ˙Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ˙Supported: replaces, timer ˙Content-Length: 0 ˙ ˙ ˙<------------> [Mar 21 17:24:40] VERBOSE[10297][C-00000001] bridge_channel.c: -- Channel SIP/6003-00000003 left 'simple_bridge' basic-bridge [Mar 21 17:24:40] VERBOSE[10294][C-00000000] bridge_channel.c: -- Channel SIP/6001-00000000 left 'simple_bridge' basic-bridge [Mar 21 17:24:40] VERBOSE[10294][C-00000000] pbx.c: == Spawn extension (from-internal, 6002, 1) exited non-zero on 'SIP/6001-00000000' [Mar 21 17:24:40] VERBOSE[10294][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' in 32000 ms (Method: ACK) [Mar 21 17:24:40] VERBOSE[10294][C-00000000] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 21 17:24:40] VERBOSE[10294][C-00000000] chan_sip.c: set_destination: set destination to 10.24.18.16:5060 [Mar 21 17:24:40] VERBOSE[10294][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 10.24.18.16:5060: ˙BYE sip:6001@10.24.18.16:5060;ob SIP/2.0 ˙Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6d9ee855;rport ˙Max-Forwards: 70 ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙CSeq: 105 BYE ˙User-Agent: Asterisk PBX SVN-branch-12-r410933 ˙Proxy-Authorization: Digest username="6001", realm="asterisk", algorithm=MD5, uri="sip:10.24.18.124", nonce="208379fb", response="6b51a04f5e2226188dd5e3927aff1850" ˙X-Asterisk-HangupCause: Normal Clearing ˙X-Asterisk-HangupCauseCode: 16 ˙Content-Length: 0 ˙ ˙ ˙--- [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: ˙<--- SIP read from UDP:10.24.18.16:5060 ---> ˙SIP/2.0 200 OK ˙Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK6d9ee855 ˙Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ ˙From: ;tag=as28bb3d0e ˙To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m ˙CSeq: 105 BYE ˙Content-Length: 0 ˙ ˙<-------------> [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' Method: ACK [Mar 21 17:24:41] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '1096126b-1194f96a-fd46e4ad@10.24.18.180' Method: REGISTER [Mar 21 17:24:48] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' Method: BYE [Mar 21 17:24:57] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' Method: REGISTER