[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] DEBUG[10291] config.c: Parsing /etc/asterisk/logger.conf [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjf6ouUHIpfS7Qnz3ea96WMSRhzNap27FC [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 6 [ 20]: CSeq: 35642 REGISTER [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 8 [ 50]: Contact: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 300 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: = Looking for Call ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV (Checking From) --From tag 22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH --To-tag [Mar 21 17:23:53] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for m6qybWzN.pQsFuItAEreYfCYc7g34OiV - REGISTER (No RTP) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjp6QWZY61f-cZAtFc6QZnXXHoRW08emya [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 6 [ 20]: CSeq: 35643 REGISTER [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 8 [ 50]: Contact: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 300 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 10 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 11 [162]: Authorization: Digest username="6001", realm="asterisk", nonce="0ea88942", uri="sip:10.24.18.124:5060", response="0d9a4cf998fecb18a1dc1aeada67936d", algorithm=MD5 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: = Looking for Call ID: m6qybWzN.pQsFuItAEreYfCYc7g34OiV (Checking From) --From tag 22ZyRDcDFKnBk0ZeY0ZcuWvy7I705wLH --To-tag [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid m6qybWzN.pQsFuItAEreYfCYc7g34OiV [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Store REGISTER's Contact header for call routing. [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: build_path: do not use Path headers [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:23:53] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:23:53] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:23:53] DEBUG[10288] app_queue.c: Device 'SIP/6001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjNfwx-c7QwuGq84MDyveExyrXTac9WtHF [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 7 [ 21]: CSeq: 28240 SUBSCRIBE [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 8 [ 22]: Event: message-summary [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 9 [ 13]: Expires: 3600 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: = Looking for Call ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU (Checking From) --From tag xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW --To-tag [Mar 21 17:23:53] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for EfpQTvraASBU5bJNxdvadNsbie4hQPdU - SUBSCRIBE (No RTP) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid EfpQTvraASBU5bJNxdvadNsbie4hQPdU [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: build_route: Contact hop: "RustyONE" [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjrsgb9myTX73MzE5HMv.anTfgLpzDm37V [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 4 [ 25]: To: sip:6001@10.24.18.124 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 7 [ 21]: CSeq: 13230 SUBSCRIBE [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 8 [ 15]: Event: presence [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 600 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 11 [ 51]: Accept: application/pidf+xml, application/xpidf+xml [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (15 headers 0 lines) --- [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: = Looking for Call ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh (Checking From) --From tag bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H --To-tag [Mar 21 17:23:53] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh - SUBSCRIBE (No RTP) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: build_route: Contact hop: "RustyONE" [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjinQ0qrKMjxH9EkGyIwsrfPukKgeuGue3 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 4 [ 38]: To: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 7 [ 21]: CSeq: 28241 SUBSCRIBE [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 8 [ 22]: Event: message-summary [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 9 [ 13]: Expires: 3600 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 11 [ 42]: Accept: application/simple-message-summary [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 14 [167]: Authorization: Digest username="6001", realm="asterisk", nonce="01cd83e3", uri="sip:6001@10.24.18.124:5060", response="700a90c1b9948867dbcfe4d9266ea1fd", algorithm=MD5 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: = Looking for Call ID: EfpQTvraASBU5bJNxdvadNsbie4hQPdU (Checking From) --From tag xm5f2qIXkVoJUWFddTTZmJ1E-V3JEvSW --To-tag [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Got a new subscription EfpQTvraASBU5bJNxdvadNsbie4hQPdU (possibly with auth) or retransmission [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid EfpQTvraASBU5bJNxdvadNsbie4hQPdU [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: build_route: Retaining previous route: [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:23:53] NOTICE[10274] chan_sip.c: Received SIP subscribe for peer without mailbox: 6001 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Destroying SIP dialog EfpQTvraASBU5bJNxdvadNsbie4hQPdU [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] DEBUG[10274] chan_sip.c: Header 0 [ 44]: SUBSCRIBE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPje49HN9pC1NWMm2Iknk31s1xzDwE0gqU4 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 4 [ 25]: To: sip:6001@10.24.18.124 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 7 [ 21]: CSeq: 13231 SUBSCRIBE [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 8 [ 15]: Event: presence [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 600 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 11 [ 51]: Accept: application/pidf+xml, application/xpidf+xml [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 13 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 14 [167]: Authorization: Digest username="6001", realm="asterisk", nonce="5b448484", uri="sip:6001@10.24.18.124:5060", response="ac53da48636015c11b92d0ce0b5e8a2b", algorithm=MD5 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: --- (16 headers 0 lines) --- [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: = Looking for Call ID: ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh (Checking From) --From tag bxYWzYnG-Igy6uFJqunKdZD4-fIurj0H --To-tag [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Got a new subscription ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh (possibly with auth) or retransmission [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Creating new subscription [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:53] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: build_route: Retaining previous route: [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:53] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:53] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:23:53] DEBUG[10274] chan_sip.c: Destroying SIP dialog ubt2ZcugDdXVt1vld2mcL.9H4gJA1WCh [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6002@10.24.18.124 SIP/2.0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj76WseEzga1EaojalP5Cij5JMboutQq7Z [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 20917 INVITE [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 14 [ 19]: Content-Length: 288 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 15 [ 0]: [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90811929 90811929 IN IP4 10.24.18.16 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4020 RTP/AVP 0 8 9 96 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4021 IN IP4 10.24.18.16 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (15 headers 14 lines) --- [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking From) --From tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m --To-tag [Mar 21 17:23:57] DEBUG[10274] acl.c: For destination '10.24.18.16', our source address is '10.24.18.124'. [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:23:57] DEBUG[10274] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:57] DEBUG[10274] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.16:5060 (no NAT) [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ - INVITE (No RTP) [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: timer [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 21 17:23:57] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: ACK sip:6002@10.24.18.124 SIP/2.0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj76WseEzga1EaojalP5Cij5JMboutQq7Z [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as7333f32b [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 20917 ACK [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking From) --From tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m --To-tag as7333f32b [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' of Response 20917: Match Found [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6002@10.24.18.124 SIP/2.0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPj-8x1t4VTHr0yC-A26sPV4zQbT5GWbIDi [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 5 [ 50]: Contact: "RustyONE" [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 20918 INVITE [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 13 [162]: Authorization: Digest username="6001", realm="asterisk", nonce="208379fb", uri="sip:6002@10.24.18.124", response="005d7cb521c26e6ee7ee44d797ac65ac", algorithm=MD5 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 15 [ 19]: Content-Length: 288 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 16 [ 0]: [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90811929 90811929 IN IP4 10.24.18.16 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4020 RTP/AVP 0 8 9 96 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4021 IN IP4 10.24.18.16 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking From) --From tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m --To-tag [Mar 21 17:23:57] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:57] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5d6802bae8' [Mar 21 17:23:57] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Allocated port 18866 for RTP instance '0x7f5d6802bae8' [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: RTP instance '0x7f5d6802bae8' is setup and ready to go [Mar 21 17:23:57] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802bae8' [Mar 21 17:23:57] VERBOSE[10274][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811929 IN IP4 10.24.18.16... OK. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f8590 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 8 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0x7f5d818f8590 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 9 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 9 based on m type on 0x7f5d818f8590 [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f8590 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4021 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6802bae8' [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f8590 to 0x7f5d6802bc28 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 8 from 0x7f5d818f8590 to 0x7f5d6802bc28 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 9 from 0x7f5d818f8590 to 0x7f5d6802bc28 [Mar 21 17:23:57] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f8590 to 0x7f5d6802bc28 [Mar 21 17:23:57] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6802bae8' [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Checking SIP call limits for device 6001 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for incoming call [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: INVITE also has "Min-SE" header. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Received Min-SE: 90 [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: build_route: Contact hop: "RustyONE" [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: Session timer started: 24 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ 900000ms [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: SIP/6001-00000000: New call is still down.... Trying... [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:23:57] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:23:57] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:23:57] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:23:57] DEBUG[10294][C-00000000] pbx.c: Result of 'EXTEN' is '6002' [Mar 21 17:23:57] DEBUG[10294][C-00000000] pbx.c: Launching 'Dial' [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] DEBUG[10294][C-00000000] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Allocating new SIP dialog for 3fb5e8e237043d3d16d84150638c0bf8@127.0.1.1:5060 - INVITE (No RTP) [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5dc800b4f8' [Mar 21 17:23:57] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Allocated port 15552 for RTP instance '0x7f5dc800b4f8' [Mar 21 17:23:57] DEBUG[10294][C-00000000] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:23:57] DEBUG[10294][C-00000000] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: RTP instance '0x7f5dc800b4f8' is setup and ready to go [Mar 21 17:23:57] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b4f8' [Mar 21 17:23:57] VERBOSE[10294][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Mar 21 17:23:57] DEBUG[10294][C-00000000] acl.c: For destination '10.24.18.138', our source address is '10.24.18.124'. [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: SIP call-id changed from '3fb5e8e237043d3d16d84150638c0bf8@127.0.1.1:5060' to '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:23:57] DEBUG[10294][C-00000000] channel_internal_api.c: Channel Call ID changing from [C-00000000] to [C-00000000] [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d6802bc28 to 0x7f5dc800b638 [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Copying payload 8 from 0x7f5d6802bc28 to 0x7f5dc800b638 [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Copying payload 9 from 0x7f5d6802bc28 to 0x7f5dc800b638 [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d6802bc28 to 0x7f5dc800b638 [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Seeded SDP of 'SIP/6002-00000001' with that of 'SIP/6001-00000000' [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Outgoing Call for 6002 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10294][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c404372 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 4 [ 35]: To: [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 9 [ 35]: Date: Fri, 21 Mar 2014 22:23:57 GMT [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10294][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #26 [Mar 21 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:23:57] VERBOSE[10294][C-00000000] app_dial.c: -- Called SIP/6002 [Mar 21 17:23:57] DEBUG[10294][C-00000000] channel.c: SIP/6001-00000000: Dropping redundant connected line update "Bob" <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] DEBUG[10274] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK6c404372 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 4 [ 30]: To: [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking To) --From tag as1ecd7421 --To-tag [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Cancelling retransmission #26 - INVITE (got response) [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' Request 102: Found [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: SIP response 100 to standard invite [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] DEBUG[10274] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 ringing [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK6c404372 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 8 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:23:57] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:23:57] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking To) --From tag as1ecd7421 --To-tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' Request 102: Found [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: SIP response 180 to standard invite [Mar 21 17:23:57] DEBUG[10274][C-00000000] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:23:57] VERBOSE[10274][C-00000000] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:57] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:23:57] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:23:57] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:23:57] VERBOSE[10294][C-00000000] app_dial.c: -- SIP/6002-00000001 is ringing [Mar 21 17:23:57] DEBUG[10294][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/6001-00000000' with that of 'SIP/6002-00000001' [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 17:23:57] DEBUG[10294][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:23:59] DEBUG[10294][C-00000000] res_rtp_asterisk.c: 0x7f5dc800fa40 -- Probation learning mode pass with source address 10.24.18.138:4010 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK6c404372 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 8 [ 51]: Contact: "RustyTWO" [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 10 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 243 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811929 90811930 IN IP4 10.24.18.138 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4010 RTP/AVP 0 96 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4011 IN IP4 10.24.18.138 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:23:59] VERBOSE[10274] chan_sip.c: --- (13 headers 12 lines) --- [Mar 21 17:23:59] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking To) --From tag as1ecd7421 --To-tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Acked pending invite 102 [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' of Request 102: Match Found [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: SIP response 200 to standard invite [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811930 IN IP4 10.24.18.138... OK. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:23:59] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:23:59] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:23:59] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:23:59] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4011 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b4f8' [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dc800b638 [Mar 21 17:23:59] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dc800b638 [Mar 21 17:23:59] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dc800b4f8' [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:23:59] VERBOSE[10274][C-00000000] chan_sip.c: list_route: route/path hop: [Mar 21 17:23:59] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:23:59] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:23:59] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:23:59] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:23:59] DEBUG[10294][C-00000000] channel.c: SIP/6001-00000000: Dropping redundant connected line update "Bob" <6002>. [Mar 21 17:23:59] VERBOSE[10294][C-00000000] app_dial.c: -- SIP/6002-00000001 answered SIP/6001-00000000 [Mar 21 17:23:59] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:23:59] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:23:59] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:23:59] DEBUG[10294][C-00000000] rtp_engine.c: Setting early bridge SDP of 'SIP/6001-00000000' with that of 'SIP/6002-00000001' [Mar 21 17:23:59] DEBUG[10294][C-00000000] chan_sip.c: SIP answering channel: SIP/6001-00000000 [Mar 21 17:23:59] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:23:59] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:23:59] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:23:59] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:23:59] DEBUG[10294][C-00000000] chan_sip.c: Setting framing from config on incoming call [Mar 21 17:23:59] DEBUG[10294][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:23:59] DEBUG[10294][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10294][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:23:59] DEBUG[10294][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [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] DEBUG[10294][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #29 [Mar 21 17:23:59] DEBUG[10294][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:23:59] DEBUG[10294][C-00000000] features.c: Removing dialed interfaces datastore on SIP/6002-00000001 since we're bridging [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge_native_rtp.c: Bridge '48397e75-23cd-41a2-9a69-6d056da1192b' can not use native RTP bridge as two channels are required [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology constructor [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology start [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8019908(SIP/6001-00000000) is joining [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8016dd8(SIP/6002-00000001) is joining [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: pushing 0x7f5dc8019908(SIP/6001-00000000) [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] DEBUG[10294][C-00000000] bridge_native_rtp.c: Bridge '48397e75-23cd-41a2-9a69-6d056da1192b' can not use native RTP bridge as two channels are required [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is already using the new technology. [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6001-00000000 already has read format ulaw [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6001-00000000 already has write format ulaw [Mar 21 17:23:59] DEBUG[10294][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8019908(SIP/6001-00000000) is joining simple_bridge technology [Mar 21 17:23:59] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: pushing 0x7f5dc8016dd8(SIP/6002-00000001) [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] DEBUG[10242] cdr.c: Finalized CDR for SIP/6002-00000001 - start 1395440637.507266 answer 1395440639.934400 end 1395440639.936103 dispo ANSWERED [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Chose bridge technology native_rtp [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] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling native_rtp technology constructor [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology stop [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8019908(SIP/6001-00000000) is leaving simple_bridge technology (dummy) [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6001-00000000 already has read format ulaw [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6001-00000000 already has write format ulaw [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8019908(SIP/6001-00000000) is joining native_rtp technology [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Deferring reinvite on SIP 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' - It's audio will be redirected to IP 10.24.18.138:4010 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Sending reinvite on SIP '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.16:4020 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Strict routing enforced for session 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [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] DEBUG[10295][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:23:59] DEBUG[10295][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10295][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10295][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK45fe1d4e [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 4 [ 72]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10295][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #30 [Mar 21 17:23:59] DEBUG[10295][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge_native_rtp.c: Remotely bridged 'SIP/6001-00000000' and 'SIP/6002-00000001' - media will flow directly between them [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6002-00000001 already has read format ulaw [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6002-00000001 already has write format ulaw [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8016dd8(SIP/6002-00000001) is joining native_rtp technology [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge_native_rtp.c: Remotely bridged 'SIP/6001-00000000' and 'SIP/6002-00000001' - media will flow directly between them [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling native_rtp technology start [Mar 21 17:23:59] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology destructor [Mar 21 17:23:59] DEBUG[10295][C-00000000] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:23:59] DEBUG[10295][C-00000000] res_rtp_asterisk.c: 0x7f5dc800fa40 -- Probation learning mode pass with source address 10.24.18.138:4010 [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 17:23:59] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:23:59] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Mar 21 17:24:00] DEBUG[10294][C-00000000] res_rtp_asterisk.c: 0x7f5d68030030 -- Probation learning mode pass with source address 10.24.18.16:4020 [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] DEBUG[10295][C-00000000] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:24:00] DEBUG[10295][C-00000000] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: ACK sip:6002@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 1 [ 88]: Via: SIP/2.0/UDP 10.24.18.16:5060;rport;branch=z9hG4bKPjvbR6GkgAYHFpM0imQ4GRtQV4C8bhmKgH [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as28bb3d0e [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 20918 ACK [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking From) --From tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m --To-tag as28bb3d0e [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' of Response 20918: Match Found [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Sending pending reinvite on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10274][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Initializing already initialized SIP dialog Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (presumably reinvite) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 0 [ 43]: INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK494166cd;rport [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 6 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #31 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK45fe1d4e [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Content-Length: 243 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 11 [ 0]: [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811929 90811931 IN IP4 10.24.18.138 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4010 RTP/AVP 0 96 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4011 IN IP4 10.24.18.138 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking To) --From tag as1ecd7421 --To-tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Acked pending invite 103 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #30 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' of Request 103: Match Found [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811931 IN IP4 10.24.18.138... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4011 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dc800b638 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dc800b638 [Mar 21 17:24:00] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b4f8' [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Destroying SIP dialog 21608363712124-1345167932089@10.24.18.166 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK494166cd [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 7 [ 50]: Contact: "RustyONE" [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 240 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90811929 90811930 IN IP4 10.24.18.16 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4020 RTP/AVP 0 96 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4021 IN IP4 10.24.18.16 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking To) --From tag as28bb3d0e --To-tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Acked pending invite 102 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' of Request 102: Match Found [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811930 IN IP4 10.24.18.16... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4021 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6802bc28 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6802bc28 [Mar 21 17:24:00] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802bae8' [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for incoming call [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Refresher: UAC [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Session timer stopped: 24 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Session timer started: 32 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ 900000ms [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Sending reinvite on SIP '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.16:4020 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Strict routing enforced for session 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [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] DEBUG[10295][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:00] DEBUG[10295][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10295][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10295][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK54805a0f [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 4 [ 72]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10295][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #33 [Mar 21 17:24:00] DEBUG[10295][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 1 [ 79]: Via: SIP/2.0/UDP 10.24.18.124:5060;received=10.24.18.124;branch=z9hG4bK54805a0f [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Content-Length: 243 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Header 11 [ 0]: [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811929 90811932 IN IP4 10.24.18.138 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4010 RTP/AVP 0 96 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4011 IN IP4 10.24.18.138 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:00] VERBOSE[10274] chan_sip.c: --- (11 headers 12 lines) --- [Mar 21 17:24:00] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking To) --From tag as1ecd7421 --To-tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Acked pending invite 104 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' of Request 104: Match Found [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811932 IN IP4 10.24.18.138... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:00] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4011 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dc800b638 [Mar 21 17:24:00] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dc800b638 [Mar 21 17:24:00] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b4f8' [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:00] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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 17:24:00] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274] chan_sip.c: Header 0 [ 41]: INVITE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjKaKBJBea7CyKmCFJZ5EuVktXMAeYGyMt [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 6 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 14655 INVITE [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 14 [ 19]: Content-Length: 303 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 15 [ 0]: [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811929 90811933 IN IP4 10.24.18.138 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendonly [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 7 [ 29]: m=audio 4010 RTP/AVP 0 8 9 96 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 8 [ 31]: a=rtcp:4011 IN IP4 10.24.18.138 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 11 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 14 [ 10]: a=sendonly [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: --- (15 headers 15 lines) --- [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking From) --From tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV --To-tag as1ecd7421 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: timer [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 21 17:24:07] DEBUG[10274][C-00000000] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811933 IN IP4 10.24.18.138... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=sendonly... OK. [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f8590 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 8 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0x7f5d818f8590 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 9 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 9 based on m type on 0x7f5d818f8590 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f8590 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4011 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [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] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b4f8' [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f8590 to 0x7f5dc800b638 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 8 from 0x7f5d818f8590 to 0x7f5dc800b638 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 9 from 0x7f5d818f8590 to 0x7f5dc800b638 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f8590 to 0x7f5dc800b638 [Mar 21 17:24:07] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dc800b4f8' [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:07] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b4f8' [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Got a SIP re-invite for call 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: INVITE also has "Min-SE" header. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Received Min-SE: 90 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: SIP/6002-00000001: This call is UP.... [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Setting framing from config on incoming call [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10274][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #34 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802bae8' [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Sending reinvite on SIP 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' - It's audio soon redirected to IP 10.24.18.124:5060 [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10294][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:07] DEBUG[10294][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10294][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10294][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Initializing already initialized SIP dialog Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (presumably reinvite) [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 0 [ 43]: INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK66296f91;rport [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 6 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10294][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #35 [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:24:07] DEBUG[10294][C-00000000] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/6001-00000000' and 'SIP/6002-00000001' - media will flow through Asterisk core [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [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] DEBUG[10294][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:24:07] DEBUG[10294][C-00000000] chan_sip.c: Deferring reinvite on SIP 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' - It's audio will be redirected to IP (null) [Mar 21 17:24:07] DEBUG[10294][C-00000000] channel.c: Set channel SIP/6001-00000000 to write format slin [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_musiconhold.c: SIP/6001-00000000 Opened file 0 '/var/lib/asterisk/moh/macroform-cold_day' [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Difference is 58272, ms is 7304 [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: ACK sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjwRlymbhpLreWlq3LISobkELSLLRRe-lZ [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 14655 ACK [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking From) --From tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV --To-tag as1ecd7421 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' of Response 14655: Match Found [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: 0x7f5d68030030 -- Probation learning mode pass with source address 10.24.18.16:4020 [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] DEBUG[10294][C-00000000] channel.c: Generator got voice, switching to phase locked mode [Mar 21 17:24:07] DEBUG[10294][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: Changing ssrc from 869980550 to 1666408576 due to a source change [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK66296f91 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 7 [ 50]: Contact: "RustyONE" [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 240 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90811929 90811931 IN IP4 10.24.18.16 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4020 RTP/AVP 0 96 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4021 IN IP4 10.24.18.16 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:07] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:07] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking To) --From tag as28bb3d0e --To-tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Acked pending invite 103 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' of Request 103: Match Found [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811931 IN IP4 10.24.18.16... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:07] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4021 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6802bc28 [Mar 21 17:24:07] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6802bc28 [Mar 21 17:24:07] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6802bae8' [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for incoming call [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Refresher: UAC [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Session timer stopped: 32 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Session timer started: 36 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ 900000ms [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Sending pending reinvite on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:07] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10274][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Initializing already initialized SIP dialog Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (presumably reinvite) [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 0 [ 43]: INVITE sip:6001@10.24.18.16:5060;ob SIP/2.0 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK05a16f19;rport [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 6 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #37 [Mar 21 17:24:07] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: 0x7f5d68030030 -- Probation learning mode pass with source address 10.24.18.16:4020 [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] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f5d6802bae8' [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Mar 21 17:24:07] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Difference is 1752, ms is 239 [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: Changing ssrc from 1666408576 to 290772981 due to a source change [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:07] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK05a16f19 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 7 [ 50]: Contact: "RustyONE" [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 240 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 1 [ 40]: o=- 90811929 90811932 IN IP4 10.24.18.16 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 3 [ 20]: c=IN IP4 10.24.18.16 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4020 RTP/AVP 0 96 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 7 [ 30]: a=rtcp:4021 IN IP4 10.24.18.16 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:08] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking To) --From tag as28bb3d0e --To-tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Acked pending invite 104 [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #37 [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' of Request 104: Match Found [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: SIP response 200 to RE-invite on outgoing call Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP o=- 90811929 90811932 IN IP4 10.24.18.16... OK. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:08] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16' into... [Mar 21 17:24:08] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port ''. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.16... OK. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:08] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:08] VERBOSE[10274][C-00000000] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:08] DEBUG[10274][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4021 IN IP4 10.24.18.16... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6802bae8' [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] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6802bc28 [Mar 21 17:24:08] DEBUG[10274][C-00000000] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6802bc28 [Mar 21 17:24:08] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6802bae8' [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Updating call counter for incoming call [Mar 21 17:24:08] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:08] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Refresher: UAC [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Session timer stopped: 36 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Session timer started: 39 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ 900000ms [Mar 21 17:24:08] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:08] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:24:08] DEBUG[10294][C-00000000] res_rtp_asterisk.c: 0x7f5d68030030 -- Probation learning mode pass with source address 10.24.18.16:4020 [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] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog '8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL' [Mar 21 17:24:08] DEBUG[10274] chan_sip.c: Destroying SIP dialog 8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL [Mar 21 17:24:08] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '8J8et6Xt9DldqJYkyPSyDNIs51Mgx8eL' Method: REGISTER [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:08] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bK5fd638b9FCFDB300 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 2 [ 58]: From: "6003" ;tag=416D771C-70EB3F17 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 3 [ 27]: To: [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 1 REGISTER [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 5 [ 48]: Call-ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 6 [137]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 10 [ 12]: Expires: 300 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: = Looking for Call ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 (Checking From) --From tag 416D771C-70EB3F17 --To-tag [Mar 21 17:24:09] DEBUG[10274] acl.c: For destination '10.24.18.180', our source address is '10.24.18.124'. [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for 1096126b-1194f96a-fd46e4ad@10.24.18.180 - REGISTER (No RTP) [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 1096126b-1194f96a-fd46e4ad@10.24.18.180 [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.180:5060 [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] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: REGISTER sip:10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bK4653224C3B24BFF [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 2 [ 58]: From: "6003" ;tag=416D771C-70EB3F17 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 3 [ 27]: To: [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 2 REGISTER [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 5 [ 48]: Call-ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 6 [137]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 9 [162]: Authorization: Digest username="6003", realm="asterisk", nonce="30c935b5", uri="sip:10.24.18.124:5060", response="7bc4c5112001ed8978a8f017501a13bd", algorithm=MD5 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 11 [ 12]: Expires: 300 [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: = Looking for Call ID: 1096126b-1194f96a-fd46e4ad@10.24.18.180 (Checking From) --From tag 416D771C-70EB3F17 --To-tag [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port '5060'. [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 1096126b-1194f96a-fd46e4ad@10.24.18.180 [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:09] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.180:5060 (no NAT) [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: Store REGISTER's Contact header for call routing. [Mar 21 17:24:09] DEBUG[10274] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:09] DEBUG[10274] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:09] DEBUG[10274] chan_sip.c: build_path: do not use Path headers [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.180:5060 [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:24:09] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:24:09] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:24:09] DEBUG[10288] app_queue.c: Device 'SIP/6003' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:09] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6003@10.24.18.124 SIP/2.0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjqdYZ6JhR5.byBIcxJhzkK68-39SU.nf6 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 16872 INVITE [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 13 [ 37]: X-Digium-Call-Hint: potentialTransfer [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 15 [ 19]: Content-Length: 291 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 16 [ 0]: [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811942 90811942 IN IP4 10.24.18.138 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4012 RTP/AVP 0 8 9 96 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4013 IN IP4 10.24.18.138 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (16 headers 14 lines) --- [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking From) --From tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx --To-tag [Mar 21 17:24:10] DEBUG[10274] acl.c: For destination '10.24.18.138', our source address is '10.24.18.124'. [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:24:10] DEBUG[10274] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:10] DEBUG[10274] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.138:5060 (no NAT) [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 - INVITE (No RTP) [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Found SIP option: -replaces- [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Matched SIP option: replaces [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Found SIP option: -100rel- [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Matched SIP option: 100rel [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Found SIP option: -timer- [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Matched SIP option: timer [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Found SIP option: -norefersub- [Mar 21 17:24:10] DEBUG[10274][C-00000001] sip/reqresp_parser.c: Matched SIP option: norefersub [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #43 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: ACK sip:6003@10.24.18.124 SIP/2.0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjqdYZ6JhR5.byBIcxJhzkK68-39SU.nf6 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as0f822371 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 16872 ACK [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking From) --From tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx --To-tag as0f822371 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' of Response 16872: Match Found [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 36]: INVITE sip:6003@10.24.18.124 SIP/2.0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPj1UIVnvufifyEK6zN5Fl2reID84W3wb54 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 4 [ 27]: To: [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 6 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 7 [ 18]: CSeq: 16873 INVITE [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 10 [ 21]: Session-Expires: 1800 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 12 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 13 [ 37]: X-Digium-Call-Hint: potentialTransfer [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 14 [162]: Authorization: Digest username="6002", realm="asterisk", nonce="208aa590", uri="sip:6003@10.24.18.124", response="be1414879cdfc964aac84a1d0096b286", algorithm=MD5 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 16 [ 19]: Content-Length: 291 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 17 [ 0]: [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811942 90811942 IN IP4 10.24.18.138 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 6 [ 29]: m=audio 4012 RTP/AVP 0 8 9 96 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4013 IN IP4 10.24.18.138 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 10 [ 20]: a=rtpmap:9 G722/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 12 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Body 13 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (17 headers 14 lines) --- [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking From) --From tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx --To-tag [Mar 21 17:24:10] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:10] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5d6803ba58' [Mar 21 17:24:10] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Allocated port 19846 for RTP instance '0x7f5d6803ba58' [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: RTP instance '0x7f5d6803ba58' is setup and ready to go [Mar 21 17:24:10] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6803ba58' [Mar 21 17:24:10] VERBOSE[10274][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 90811942 90811942 IN IP4 10.24.18.138... OK. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f8590 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 8 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 8 based on m type on 0x7f5d818f8590 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 9 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 9 based on m type on 0x7f5d818f8590 [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f8590 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6803ba58' [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f8590 to 0x7f5d6803bb98 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 8 from 0x7f5d818f8590 to 0x7f5d6803bb98 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 9 from 0x7f5d818f8590 to 0x7f5d6803bb98 [Mar 21 17:24:10] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f8590 to 0x7f5d6803bb98 [Mar 21 17:24:10] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6803ba58' [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Checking SIP call limits for device 6002 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for incoming call [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274][C-00000001] chan_sip.c: Incoming INVITE with 'timer' option supported [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: INVITE also has "Session-Expires" header. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: INVITE also has "Min-SE" header. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Received Min-SE: 90 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: build_route: Contact hop: "RustyTWO" [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: list_route: route/path hop: [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: Session timer started: 45 - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 900000ms [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: SIP/6002-00000002: New call is still down.... Trying... [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:10] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:24:10] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:24:10] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:24:10] DEBUG[10296][C-00000001] pbx.c: Result of 'EXTEN' is '6003' [Mar 21 17:24:10] DEBUG[10296][C-00000001] pbx.c: Launching 'Dial' [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] DEBUG[10296][C-00000001] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Allocating new SIP dialog for 425dd70908cd0a2a4139bcce25f2406b@127.0.1.1:5060 - INVITE (No RTP) [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f5dd000b548' [Mar 21 17:24:10] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Allocated port 19470 for RTP instance '0x7f5dd000b548' [Mar 21 17:24:10] DEBUG[10296][C-00000001] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:10] DEBUG[10296][C-00000001] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: RTP instance '0x7f5dd000b548' is setup and ready to go [Mar 21 17:24:10] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd000b548' [Mar 21 17:24:10] VERBOSE[10296][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Mar 21 17:24:10] DEBUG[10296][C-00000001] acl.c: For destination '10.24.18.180', our source address is '10.24.18.124'. [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: SIP call-id changed from '425dd70908cd0a2a4139bcce25f2406b@127.0.1.1:5060' to '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: *** Our native formats are (ulaw) [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: *** Joint capabilities are (ulaw) [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: *** Our capabilities are (ulaw) [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: This channel will not be able to handle video. [Mar 21 17:24:10] DEBUG[10296][C-00000001] channel_internal_api.c: Channel Call ID changing from [C-00000001] to [C-00000001] [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d6803bb98 to 0x7f5dd000b688 [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Copying payload 8 from 0x7f5d6803bb98 to 0x7f5dd000b688 [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Copying payload 9 from 0x7f5d6803bb98 to 0x7f5dd000b688 [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d6803bb98 to 0x7f5dd000b688 [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Seeded SDP of 'SIP/6003-00000003' with that of 'SIP/6002-00000002' [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Outgoing Call for 6003 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10296][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 4 [ 32]: To: [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 6 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 9 [ 35]: Date: Fri, 21 Mar 2014 22:24:10 GMT [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10296][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #47 [Mar 21 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:10] VERBOSE[10296][C-00000001] app_dial.c: -- Called SIP/6003 [Mar 21 17:24:10] DEBUG[10296][C-00000001] channel.c: SIP/6002-00000002: Dropping redundant connected line update "Charlie" <6003>. [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: *** SIP TIMER: Cancelling retransmission #47 - INVITE (got response) [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' Request 102: Found [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: SIP response 100 to standard invite [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 8 [ 34]: Allow-Events: talk,hold,conference [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 9 [ 19]: Accept-Language: en [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Mar 21 17:24:10] VERBOSE[10274] chan_sip.c: --- (11 headers 0 lines) --- [Mar 21 17:24:10] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' Request 102: Found [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: SIP response 180 to standard invite [Mar 21 17:24:10] DEBUG[10274][C-00000001] chan_sip.c: build_route: Contact hop: [Mar 21 17:24:10] VERBOSE[10274][C-00000001] chan_sip.c: list_route: route/path hop: [Mar 21 17:24:10] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:24:10] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:24:10] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:24:10] VERBOSE[10296][C-00000001] app_dial.c: -- SIP/6003-00000003 is ringing [Mar 21 17:24:10] DEBUG[10296][C-00000001] rtp_engine.c: Setting early bridge SDP of 'SIP/6002-00000002' with that of 'SIP/6003-00000003' [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 17:24:10] DEBUG[10296][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:10] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:11] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK6c336516 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440649 1395440649 IN IP4 10.24.18.180 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2224 RTP/AVP 0 96 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 102 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' of Request 102: Match Found [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to standard invite [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 1395440649 1395440649 IN IP4 10.24.18.180... OK. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:12] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:12] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd000b548' [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:12] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:12] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dd000b548' [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: build_route: Contact hop: [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: list_route: route/path hop: [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:12] DEBUG[10296][C-00000001] channel.c: SIP/6002-00000002: Dropping redundant connected line update "Charlie" <6003>. [Mar 21 17:24:12] VERBOSE[10296][C-00000001] app_dial.c: -- SIP/6003-00000003 answered SIP/6002-00000002 [Mar 21 17:24:12] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:24:12] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:24:12] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:24:12] DEBUG[10296][C-00000001] rtp_engine.c: Setting early bridge SDP of 'SIP/6002-00000002' with that of 'SIP/6003-00000003' [Mar 21 17:24:12] DEBUG[10296][C-00000001] chan_sip.c: SIP answering channel: SIP/6002-00000002 [Mar 21 17:24:12] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:24:12] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:24:12] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:24:12] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:24:12] DEBUG[10296][C-00000001] chan_sip.c: Setting framing from config on incoming call [Mar 21 17:24:12] DEBUG[10296][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:12] DEBUG[10296][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10296][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:12] DEBUG[10296][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [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] DEBUG[10296][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #50 [Mar 21 17:24:12] DEBUG[10296][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:12] DEBUG[10296][C-00000001] features.c: Removing dialed interfaces datastore on SIP/6003-00000003 since we're bridging [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge_native_rtp.c: Bridge 'e501657b-704b-4b2f-9d8f-55487218673b' can not use native RTP bridge as two channels are required [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology constructor [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology start [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001a9f8(SIP/6002-00000002) is joining [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001bdd8(SIP/6003-00000003) is joining [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: pushing 0x7f5dd001a9f8(SIP/6002-00000002) [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] DEBUG[10296][C-00000001] bridge_native_rtp.c: Bridge 'e501657b-704b-4b2f-9d8f-55487218673b' can not use native RTP bridge as two channels are required [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is already using the new technology. [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6002-00000002 already has read format ulaw [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6002-00000002 already has write format ulaw [Mar 21 17:24:12] DEBUG[10296][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001a9f8(SIP/6002-00000002) is joining simple_bridge technology [Mar 21 17:24:12] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: pushing 0x7f5dd001bdd8(SIP/6003-00000003) [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] DEBUG[10242] cdr.c: Finalized CDR for SIP/6003-00000003 - start 1395440650.398482 answer 1395440652.906030 end 1395440652.907684 dispo ANSWERED [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Chose bridge technology native_rtp [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] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling native_rtp technology constructor [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology stop [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001a9f8(SIP/6002-00000002) is leaving simple_bridge technology (dummy) [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6002-00000002 already has read format ulaw [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6002-00000002 already has write format ulaw [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001a9f8(SIP/6002-00000002) is joining native_rtp technology [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Deferring reinvite on SIP '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' - It's audio will be redirected to IP 10.24.18.180:2224 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Sending reinvite on SIP '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.138:4012 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10297][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:12] DEBUG[10297][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10297][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10297][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK24774eda [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 4 [ 54]: To: ;tag=884188E1-2198B888 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 6 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10297][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #51 [Mar 21 17:24:12] DEBUG[10297][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge_native_rtp.c: Remotely bridged 'SIP/6002-00000002' and 'SIP/6003-00000003' - media will flow directly between them [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6003-00000003 already has read format ulaw [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6003-00000003 already has write format ulaw [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001bdd8(SIP/6003-00000003) is joining native_rtp technology [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge_native_rtp.c: Remotely bridged 'SIP/6002-00000002' and 'SIP/6003-00000003' - media will flow directly between them [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling native_rtp technology start [Mar 21 17:24:12] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology destructor [Mar 21 17:24:12] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK24774eda [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440649 1395440650 IN IP4 10.24.18.180 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2224 RTP/AVP 0 96 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:12] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 103 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #51 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' of Request 103: Match Found [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 1395440649 1395440650 IN IP4 10.24.18.180... OK. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:12] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:12] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:12] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:12] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:12] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd000b548' [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:12] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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 17:24:12] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:12] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10297][C-00000001] res_rtp_asterisk.c: 0x7f5dd000fa90 -- Probation learning mode pass with source address 10.24.18.180:2224 [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] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:24:13] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10296][C-00000001] res_rtp_asterisk.c: 0x7f5d6803ffa0 -- Probation learning mode pass with source address 10.24.18.138:4012 [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] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Mar 21 17:24:13] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Mar 21 17:24:13] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f5dd000b548' [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: ACK sip:6003@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjx8DdIatSNQEBTLVPUdcOBRO0WQZWAd.. [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 3 [ 77]: From: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 4 [ 42]: To: ;tag=as5a51f621 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 5 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 16873 ACK [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: --- (8 headers 0 lines) --- [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking From) --From tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx --To-tag as5a51f621 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #50 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' of Response 16873: Match Found [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Sending pending reinvite on '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10274][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (presumably reinvite) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3b5f4909;rport [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 3 [ 44]: From: ;tag=as5a51f621 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 6 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #53 [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK3b5f4909 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as5a51f621 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 7 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 243 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811942 90811943 IN IP4 10.24.18.138 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4012 RTP/AVP 0 96 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4013 IN IP4 10.24.18.138 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:13] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking To) --From tag as5a51f621 --To-tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 102 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #53 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' of Request 102: Match Found [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 90811942 90811943 IN IP4 10.24.18.138... OK. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:13] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5d6803bb98 [Mar 21 17:24:13] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5d6803bb98 [Mar 21 17:24:13] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6803ba58' [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for incoming call [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Refresher: UAC [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Session timer stopped: 45 - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Session timer started: 54 - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 900000ms [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Sending reinvite on SIP '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.138:4012 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10297][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:13] DEBUG[10297][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10297][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10297][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK732a9bac [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 4 [ 54]: To: ;tag=884188E1-2198B888 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 6 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10297][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #55 [Mar 21 17:24:13] DEBUG[10297][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK732a9bac [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 104 INVITE [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440649 1395440651 IN IP4 10.24.18.180 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2224 RTP/AVP 0 96 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:13] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 104 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #55 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' of Request 104: Match Found [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 1395440649 1395440651 IN IP4 10.24.18.180... OK. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:13] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:13] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:13] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:13] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd000b548' [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:13] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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 17:24:13] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:13] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:14] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: No remote address on RTP instance '0x7f5dc800b4f8' so dropping frame [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] DEBUG[10274] chan_sip.c: Header 0 [ 40]: REFER sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPjMVSpfmLL-nFUrBvMn3ZLpvDTuQbdP39j [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 5 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 6 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 7 [ 17]: CSeq: 14656 REFER [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 8 [ 12]: Event: refer [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 9 [ 12]: Expires: 600 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 11 [ 35]: Accept: message/sipfrag;version=2.0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 12 [ 46]: Allow-Events: presence, message-summary, refer [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 13 [143]: Refer-To: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 14 [ 39]: Referred-By: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 15 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (17 headers 0 lines) --- [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking From) --From tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV --To-tag as1ecd7421 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [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] DEBUG[10274][C-00000000] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 F-tag: P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx T-tag: as5a51f621 [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] DEBUG[10274][C-00000000] chan_sip.c: SIP attended transfer: Transferer channel SIP/6002-00000001 [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] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Looking for callid 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (fromtag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx totag as5a51f621) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Matched INCOMING call - their tag is P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx Our tag is as5a51f621 [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Moving 0x7f5dc8019908(SIP/6001-00000000) into bridge e501657b-704b-4b2f-9d8f-55487218673b swapping with SIP/6002-00000002 [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: pulling 0x7f5dc8019908(SIP/6001-00000000) [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] DEBUG[10274][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8019908(SIP/6001-00000000) is leaving native_rtp technology [Mar 21 17:24:15] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dc800b4f8' [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/6001-00000000' and 'SIP/6002-00000001' - media will flow through Asterisk core [Mar 21 17:24:15] DEBUG[10242] cdr.c: Finalized CDR for SIP/6001-00000000 - start 1395440637.504949 answer 1395440639.934918 end 1395440655.362081 dispo ANSWERED [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: pushing 0x7f5dc8019908(SIP/6001-00000000) by swapping with 0x7f5dd001a9f8(SIP/6002-00000002) [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] DEBUG[10274][C-00000000] bridge_channel.c: Setting 0x7f5dd001a9f8(SIP/6002-00000002) state from:0 to:2 [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: pulling 0x7f5dd001a9f8(SIP/6002-00000002) [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] DEBUG[10274][C-00000000] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001a9f8(SIP/6002-00000002) is leaving native_rtp technology [Mar 21 17:24:15] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5d6803ba58' [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Sending reinvite on SIP '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' - It's audio soon redirected to IP 10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [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] DEBUG[10274][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (presumably reinvite) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 0 [ 44]: INVITE sip:6002@10.24.18.138:5060;ob SIP/2.0 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3c619fd5;rport [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 3 [ 44]: From: ;tag=as5a51f621 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 6 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 10 [ 10]: Min-SE: 90 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 11 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 12 [ 26]: Supported: replaces, timer [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 13 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 14 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #56 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/6002-00000002' and 'SIP/6001-00000000' - media will flow through Asterisk core [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_native_rtp.c: Bridge 'e501657b-704b-4b2f-9d8f-55487218673b' can not use native RTP bridge as channel 'SIP/6003-00000003' has features which prevent it [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Chose bridge technology simple_bridge [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] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology constructor [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling native_rtp technology stop [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001bdd8(SIP/6003-00000003) is leaving native_rtp technology (dummy) [Mar 21 17:24:15] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dd000b548' [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Sending reinvite on SIP '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' - It's audio soon redirected to IP 10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) [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] DEBUG[10274][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3b85ed75 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 4 [ 54]: To: ;tag=884188E1-2198B888 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 6 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 7 [ 16]: CSeq: 105 INVITE [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #57 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6003-00000003 already has read format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6003-00000003 already has write format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001bdd8(SIP/6003-00000003) is joining simple_bridge technology [Mar 21 17:24:15] DEBUG[10274][C-00000000] channel.c: Set channel SIP/6001-00000000 to write format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] channel.c: Set channel SIP/6001-00000000 to write format slin [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6001-00000000 already has read format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is happy that channel SIP/6001-00000000 already has write format slin [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dc8019908(SIP/6001-00000000) is joining simple_bridge technology [Mar 21 17:24:15] DEBUG[10274][C-00000000] channel.c: Set channel SIP/6001-00000000 to write format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] channel.c: Set channel SIP/6001-00000000 to write format slin [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology start [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling native_rtp technology destructor [Mar 21 17:24:15] DEBUG[10242] cdr.c: Finalized CDR for SIP/6002-00000002 - start 1395440650.396055 answer 1395440652.906535 end 1395440655.364369 dispo ANSWERED [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_native_rtp.c: Bridge '48397e75-23cd-41a2-9a69-6d056da1192b' can not use native RTP bridge as two channels are required [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Chose bridge technology simple_bridge [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] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology constructor [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling native_rtp technology stop [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8016dd8(SIP/6002-00000001) is leaving native_rtp technology (dummy) [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6002-00000001 already has read format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is happy that channel SIP/6002-00000001 already has write format ulaw [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8016dd8(SIP/6002-00000001) is joining simple_bridge technology [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology start [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling native_rtp technology destructor [Mar 21 17:24:15] DEBUG[10274][C-00000000] bridge_channel.c: Setting 0x7f5dc8016dd8(SIP/6002-00000001) state from:0 to:2 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Strict routing enforced for session 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [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] DEBUG[10297][C-00000001] chan_sip.c: Deferring reinvite on SIP '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' - It's audio will be redirected to IP 10.24.18.16:4020 [Mar 21 17:24:15] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:24:15] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [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] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 17:24:15] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1880200581 to 2090517288 due to a source change [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] DEBUG[10274][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10294][C-00000000] channel.c: Set channel SIP/6001-00000000 to write format ulaw [Mar 21 17:24:15] DEBUG[10296][C-00000001] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 21 17:24:15] DEBUG[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] pbx.c: == Spawn extension (from-internal, 6003, 1) exited non-zero on 'SIP/6002-00000002' [Mar 21 17:24:15] DEBUG[10296][C-00000001] channel.c: Soft-Hanging up channel 'SIP/6002-00000002' [Mar 21 17:24:15] DEBUG[10294][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Mar 21 17:24:15] DEBUG[10296][C-00000001] channel.c: Hanging up channel 'SIP/6002-00000002' [Mar 21 17:24:15] DEBUG[10296][C-00000001] chan_sip.c: Hangup call SIP/6002-00000002, SIP callid 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] DEBUG[10296][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6803ba58' [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] DEBUG[10296][C-00000001] chan_sip.c: Session timer stopped: 54 - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:24:15] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:24:15] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:24:15] DEBUG[10242] cdr.c: Finalized CDR for SIP/6001-00000000 - start 1395440655.364323 answer 1395440655.364323 end 1395440655.365736 dispo ANSWERED [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: pulling 0x7f5dc8016dd8(SIP/6002-00000001) [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] DEBUG[10295][C-00000000] bridge_channel.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: 0x7f5dc8016dd8(SIP/6002-00000001) is leaving simple_bridge technology [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: dissolving bridge with cause 16(Normal Clearing) [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: queueing action type:13 sub:1001 [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b is dissolved, not performing smart bridge operation. [Mar 21 17:24:15] DEBUG[10295][C-00000000] res_rtp_asterisk.c: Changing ssrc from 290772981 to 1629433644 due to a source change [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: actually destroying basic bridge, nobody wants it anymore [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling basic bridge destructor [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology stop [Mar 21 17:24:15] DEBUG[10295][C-00000000] bridge.c: Bridge 48397e75-23cd-41a2-9a69-6d056da1192b: calling simple_bridge technology destructor [Mar 21 17:24:15] DEBUG[10295][C-00000000] channel.c: Hanging up channel 'SIP/6002-00000001' [Mar 21 17:24:15] DEBUG[10295][C-00000000] chan_sip.c: update_call_counter(6002) - decrement call limit counter on hangup [Mar 21 17:24:15] DEBUG[10295][C-00000000] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:15] DEBUG[10295][C-00000000] chan_sip.c: Call to peer '6002' removed from call limit 0 [Mar 21 17:24:15] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:24:15] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:24:15] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:24:15] DEBUG[10295][C-00000000] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060. [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:24:15] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:24:15] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(src) result is '6002' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(dst) result is '6003' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(6003) result is '"6003"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6002-00000002' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002-00000002) result is '"SIP/6002-00000002"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6003-00000003' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6003-00000003) result is '"SIP/6003-00000003"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6003,15' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6003,15) result is '"SIP/6003,15"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:24:10' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:10) result is '"2014-03-21 17:24:10"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:24:12' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:12) result is '"2014-03-21 17:24:12"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:24:15' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.004000' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.004000) result is '"0.004000"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.002000' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.002000) result is '"0.002000"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440650.2' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440650.2) result is '"1395440650.2"' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(sequence) result is '2' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:24:15] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:15] DEBUG[10242] cdr.c: CDR for SIP/6002-00000001 is dialed and has no Party B; discarding [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK3b85ed75 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 105 INVITE [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440649 1395440652 IN IP4 10.24.18.180 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2224 RTP/AVP 0 96 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 105 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #57 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' of Request 105: Match Found [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 1395440649 1395440652 IN IP4 10.24.18.180... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:15] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f5dd000b548' [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Sending pending reinvite on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: ** Our prefcodec: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [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] DEBUG[10274][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (presumably reinvite) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 0 [ 41]: INVITE sip:6003@10.24.18.180:5060 SIP/2.0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK47888bb4 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 3 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 4 [ 54]: To: ;tag=884188E1-2198B888 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 5 [ 37]: Contact: [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 6 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 7 [ 16]: CSeq: 106 INVITE [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 8 [ 46]: User-Agent: Asterisk PBX SVN-branch-12-r410933 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Header 12 [ 29]: Content-Type: application/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] DEBUG[10274][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #62 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:15] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f5dd000b548' [Mar 21 17:24:15] DEBUG[10297][C-00000001] res_rtp_asterisk.c: 0x7f5dd000fa90 -- Probation learning mode pass with source address 10.24.18.180:2224 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.24.18.124:5060;branch=z9hG4bK47888bb4 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 2 [ 50]: From: "Bob" ;tag=as56dbf74c [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 3 [ 61]: To: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 4 [ 16]: CSeq: 106 INVITE [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 8 [ 26]: Supported: 100rel,replaces [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 10 [ 19]: Accept-Language: en [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 12 [ 19]: Content-Length: 197 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 13 [ 0]: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 1 [ 45]: o=- 1395440649 1395440653 IN IP4 10.24.18.180 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.180 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 5 [ 25]: m=audio 2224 RTP/AVP 0 96 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 6 [ 10]: a=sendrecv [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 8 [ 32]: 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] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking To) --From tag as56dbf74c --To-tag 884188E1-2198B888 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 106 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #62 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060' of Request 106: Match Found [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 1395440649 1395440653 IN IP4 10.24.18.180... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port ''. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.180... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f79c0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [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] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd000b548' [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f79c0 to 0x7f5dd000b688 [Mar 21 17:24:15] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5dd000b548' [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:15] DEBUG[10297][C-00000001] res_rtp_asterisk.c: 0x7f5dd000fa90 -- Probation learning mode pass with source address 10.24.18.180:2224 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK3c619fd5 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as5a51f621 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 6 [ 35]: Session-Expires: 1800;refresher=uac [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 7 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 8 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 9 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 11 [ 19]: Content-Length: 243 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 12 [ 0]: [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 1 [ 41]: o=- 90811942 90811944 IN IP4 10.24.18.138 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 2 [ 8]: s=digphn [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.24.18.138 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 6 [ 25]: m=audio 4012 RTP/AVP 0 96 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 7 [ 31]: a=rtcp:4013 IN IP4 10.24.18.138 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 9 [ 10]: a=sendrecv [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 10 [ 32]: a=rtpmap:96 telephone-event/8000 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Body 11 [ 14]: a=fmtp:96 0-15 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (12 headers 12 lines) --- [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking To) --From tag as5a51f621 --To-tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Acked pending invite 103 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #56 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' of Request 103: Match Found [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP o=- 90811942 90811944 IN IP4 10.24.18.138... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port ''. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.138... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 0 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0x7f5d818f7a40 [Mar 21 17:24:15] VERBOSE[10274][C-00000001] chan_sip.c: Found RTP audio format 96 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0x7f5d818f7a40 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.138... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [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] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [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] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6803ba58' [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] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 0 from 0x7f5d818f7a40 to 0x7f5d6803bb98 [Mar 21 17:24:15] DEBUG[10274][C-00000001] rtp_engine.c: Copying payload 96 from 0x7f5d818f7a40 to 0x7f5d6803bb98 [Mar 21 17:24:15] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f5d6803ba58' [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Updating call counter for incoming call [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Session-Expires: 1800 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Refresher: UAC [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Session timer started: 64 - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 900000ms [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'ACK sip:600' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Strict routing enforced for session 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:15] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #65 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'BYE sip:600' onto UDP socket destined for 10.24.18.138:5060 [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] DEBUG[10274][C-00000001] chan_sip.c: Session timer stopped: 64 - 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK64a01688 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 2 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 3 [ 52]: From: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 4 [ 67]: To: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 105 NOTIFY [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 6 [ 51]: Contact: "RustyTWO" [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (10 headers 0 lines) --- [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking To) --From tag as1ecd7421 --To-tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #58 [Mar 21 17:24:15] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' of Request 105: Match Found [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] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK2fd19a3b [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as5a51f621 [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyTWO" ;tag=P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 5 [ 13]: CSeq: 104 BYE [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: = Looking for Call ID: 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 (Checking To) --From tag as5a51f621 --To-tag P-yktWn2iOgWS3iBNZqqkTl5XHMhh6kx [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #65 [Mar 21 17:24:15] DEBUG[10274][C-00000001] chan_sip.c: Stopping retransmission on '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' of Request 104: Match Found [Mar 21 17:24:15] DEBUG[10274] chan_sip.c: Destroying SIP dialog 698ch.tGpw6eQaut8eLL81ol9G4YdLI0 [Mar 21 17:24:15] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '698ch.tGpw6eQaut8eLL81ol9G4YdLI0' Method: ACK [Mar 21 17:24:15] DEBUG[10274] rtp_engine.c: Destroyed RTP instance '0x7f5d6803ba58' [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: BYE sip:6001@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 1 [ 89]: Via: SIP/2.0/UDP 10.24.18.138:5060;rport;branch=z9hG4bKPj8aeIXHcjCpJV9ZRdJB1TPzG8p3PwMmmU [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 3 [ 69]: From: ;tag=8PVX65lsv56p4aac9F8AU8UB9dURBSnV [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 4 [ 50]: To: "Alice" ;tag=as1ecd7421 [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 6 [ 15]: CSeq: 14657 BYE [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 7 [ 36]: User-Agent: Digium D40 1_4_0_0_57389 [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Mar 21 17:24:16] VERBOSE[10274] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 17:24:16] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 (Checking From) --From tag 8PVX65lsv56p4aac9F8AU8UB9dURBSnV --To-tag as1ecd7421 [Mar 21 17:24:16] DEBUG[10274][C-00000000] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Mar 21 17:24:16] DEBUG[10274][C-00000000] chan_sip.c: Initializing initreq for method BYE - callid 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:16] DEBUG[10274][C-00000000] netsock2.c: Splitting '10.24.18.138:5060' into... [Mar 21 17:24:16] DEBUG[10274][C-00000000] netsock2.c: ...host '10.24.18.138' and port '5060'. [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] DEBUG[10274][C-00000000] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:16] DEBUG[10274][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dc800b4f8' [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] DEBUG[10274][C-00000000] chan_sip.c: Received bye, no owner, selfdestruct soon. [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 17:24:16] DEBUG[10274][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.138:5060 [Mar 21 17:24:17] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Mar 21 17:24:22] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 40 bytes [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: REGISTER sip:10.24.18.124 SIP/2.0 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 1 [ 75]: Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK15431221533272715572;rport [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 2 [ 54]: From: 6004 ;tag=3167523177 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 3 [ 37]: To: 6004 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 4 [ 50]: Call-ID: 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 3 REGISTER [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 8 [ 11]: Expires: 60 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 9 [ 15]: Supported: path [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 10 [ 26]: User-Agent: Voip Phone 1.0 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: --- (12 headers 0 lines) --- [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: = Looking for Call ID: 21608363712124-1345167932089@10.24.18.166 (Checking From) --From tag 3167523177 --To-tag [Mar 21 17:24:25] DEBUG[10274] acl.c: For destination '10.24.18.166', our source address is '10.24.18.124'. [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.124:5060 [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Allocating new SIP dialog for 21608363712124-1345167932089@10.24.18.166 - REGISTER (No RTP) [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.24.18.166:5060 [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] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog 'm6qybWzN.pQsFuItAEreYfCYc7g34OiV' [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Destroying SIP dialog m6qybWzN.pQsFuItAEreYfCYc7g34OiV [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] DEBUG[10274] chan_sip.c: Header 0 [ 33]: REGISTER sip:10.24.18.124 SIP/2.0 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 10.24.18.166:5060;branch=z9hG4bK478224601275052328;rport [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 2 [ 54]: From: 6004 ;tag=3167523177 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 3 [ 37]: To: 6004 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 4 [ 50]: Call-ID: 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 5 [ 16]: CSeq: 4 REGISTER [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 7 [157]: Authorization: Digest username="6004", realm="asterisk", nonce="333d9384", uri="sip:10.24.18.124", response="a3a03b3588cb9a622daee27cf9fac1bb", algorithm=MD5 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 9 [ 11]: Expires: 60 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 10 [ 15]: Supported: path [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 11 [ 26]: User-Agent: Voip Phone 1.0 [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: --- (13 headers 0 lines) --- [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: = Looking for Call ID: 21608363712124-1345167932089@10.24.18.166 (Checking From) --From tag 3167523177 --To-tag [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.124' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Initializing initreq for method REGISTER - callid 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:24:25] VERBOSE[10274] chan_sip.c: Sending to 10.24.18.166:5060 (no NAT) [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.124:5060' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.124' and port ''. [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: Store REGISTER's Contact header for call routing. [Mar 21 17:24:25] DEBUG[10274] netsock2.c: Splitting '10.24.18.166:5060' into... [Mar 21 17:24:25] DEBUG[10274] netsock2.c: ...host '10.24.18.166' and port '5060'. [Mar 21 17:24:25] DEBUG[10274] chan_sip.c: build_path: do not use Path headers [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] DEBUG[10274] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.166:5060 [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] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6004 [Mar 21 17:24:25] DEBUG[10240] chan_sip.c: Checking device state for peer 6004 [Mar 21 17:24:25] DEBUG[10240] devicestate.c: Changing state for SIP/6004 - state 1 (Not in use) [Mar 21 17:24:27] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Mar 21 17:24:32] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Mar 21 17:24:37] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Got RTCP report of 40 bytes [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] DEBUG[10274] chan_sip.c: Header 0 [ 38]: BYE sip:6002@10.24.18.124:5060 SIP/2.0 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 10.24.18.180:5060;branch=z9hG4bKc5b09fd3835AF732 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 2 [ 63]: From: "6003" ;tag=884188E1-2198B888 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 3 [ 48]: To: "Bob" ;tag=as56dbf74c [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 4 [ 11]: CSeq: 1 BYE [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 5 [ 59]: Call-ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 6 [ 37]: Contact: [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: --- (11 headers 0 lines) --- [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: = Looking for Call ID: 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 (Checking From) --From tag 884188E1-2198B888 --To-tag as56dbf74c [Mar 21 17:24:40] DEBUG[10274][C-00000001] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Mar 21 17:24:40] DEBUG[10274][C-00000001] chan_sip.c: Initializing initreq for method BYE - callid 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:40] DEBUG[10274][C-00000001] netsock2.c: Splitting '10.24.18.180:5060' into... [Mar 21 17:24:40] DEBUG[10274][C-00000001] netsock2.c: ...host '10.24.18.180' and port '5060'. [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] DEBUG[10274][C-00000001] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:40] DEBUG[10274][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd000b548' [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] DEBUG[10274][C-00000001] chan_sip.c: Received bye, issuing owner hangup [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] DEBUG[10274][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.180:5060 [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge_channel.c: Setting 0x7f5dd001bdd8(SIP/6003-00000003) state from:0 to:1 [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: pulling 0x7f5dd001bdd8(SIP/6003-00000003) [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] DEBUG[10297][C-00000001] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dd001bdd8(SIP/6003-00000003) is leaving simple_bridge technology [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: dissolving bridge with cause 16(Normal Clearing) [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge_channel.c: Setting 0x7f5dc8019908(SIP/6001-00000000) state from:0 to:2 [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: queueing action type:13 sub:1001 [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is dissolved, not performing smart bridge operation. [Mar 21 17:24:40] DEBUG[10294][C-00000000] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: pulling 0x7f5dc8019908(SIP/6001-00000000) [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] DEBUG[10294][C-00000000] bridge_channel.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: 0x7f5dc8019908(SIP/6001-00000000) is leaving simple_bridge technology [Mar 21 17:24:40] DEBUG[10294][C-00000000] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b is dissolved, not performing smart bridge operation. [Mar 21 17:24:40] DEBUG[10242] cdr.c: Finalized CDR for SIP/6001-00000000 - start 1395440655.364344 answer 1395440655.364344 end 1395440680.918109 dispo ANSWERED [Mar 21 17:24:40] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Changing ssrc from 107902283 to 736551296 due to a source change [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: actually destroying basic bridge, nobody wants it anymore [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling basic bridge destructor [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology stop [Mar 21 17:24:40] DEBUG[10297][C-00000001] bridge.c: Bridge e501657b-704b-4b2f-9d8f-55487218673b: calling simple_bridge technology destructor [Mar 21 17:24:40] DEBUG[10297][C-00000001] channel.c: Hanging up channel 'SIP/6003-00000003' [Mar 21 17:24:40] DEBUG[10297][C-00000001] chan_sip.c: Hangup call SIP/6003-00000003, SIP callid 4d5142cb1ec83a7013fb6eb47f89b504@10.24.18.124:5060 [Mar 21 17:24:40] DEBUG[10297][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5dd000b548' [Mar 21 17:24:40] DEBUG[10242] cdr.c: CDR for SIP/6003-00000003 is dialed and has no Party B; discarding [Mar 21 17:24:40] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6003 [Mar 21 17:24:40] DEBUG[10240] chan_sip.c: Checking device state for peer 6003 [Mar 21 17:24:40] DEBUG[10240] devicestate.c: Changing state for SIP/6003 - state 1 (Not in use) [Mar 21 17:24:40] DEBUG[10294][C-00000000] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 21 17:24:40] DEBUG[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] pbx.c: == Spawn extension (from-internal, 6002, 1) exited non-zero on 'SIP/6001-00000000' [Mar 21 17:24:40] DEBUG[10294][C-00000000] channel.c: Soft-Hanging up channel 'SIP/6001-00000000' [Mar 21 17:24:40] DEBUG[10294][C-00000000] channel.c: Hanging up channel 'SIP/6001-00000000' [Mar 21 17:24:40] DEBUG[10294][C-00000000] chan_sip.c: Hangup call SIP/6001-00000000, SIP callid Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:40] DEBUG[10294][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f5d6802bae8' [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] DEBUG[10294][C-00000000] chan_sip.c: Session timer stopped: 39 - Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:40] DEBUG[10294][C-00000000] chan_sip.c: Strict routing enforced for session Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [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] DEBUG[10294][C-00000000] netsock2.c: Splitting '10.24.18.16:5060' into... [Mar 21 17:24:40] DEBUG[10294][C-00000000] netsock2.c: ...host '10.24.18.16' and port '5060'. [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] DEBUG[10294][C-00000000] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #73 [Mar 21 17:24:40] DEBUG[10294][C-00000000] chan_sip.c: Trying to put 'BYE sip:600' onto UDP socket destined for 10.24.18.16:5060 [Mar 21 17:24:40] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6001 [Mar 21 17:24:40] DEBUG[10240] chan_sip.c: Checking device state for peer 6001 [Mar 21 17:24:40] DEBUG[10240] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(src) result is '6001' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(6001) result is '"6001"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dst) result is '6002' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6001-00000000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6001-00000000) result is '"SIP/6001-00000000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6002-00000001' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002-00000001) result is '"SIP/6002-00000001"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6002,15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002,15) result is '"SIP/6002,15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:23:57' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:23:57) result is '"2014-03-21 17:23:57"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:23:59' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:23:59) result is '"2014-03-21 17:23:59"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:24:15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.017000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.017000) result is '"0.017000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.015000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.015000) result is '"0.015000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440637.0' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440637.0) result is '"1395440637.0"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(sequence) result is '0' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(src) result is '6001' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(6001) result is '"6001"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dst) result is '6002' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6001-00000000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6001-00000000) result is '"SIP/6001-00000000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6002-00000002' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002-00000002) result is '"SIP/6002-00000002"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6002,15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002,15) result is '"SIP/6002,15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:24:15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:24:15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:24:15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.000000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.000000) result is '"0.000000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.000000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.000000) result is '"0.000000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440637.0' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440637.0) result is '"1395440637.0"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(sequence) result is '4' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},"Destination":${CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(src) result is '6001' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(6001) result is '"6001"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},"Context":${CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dst) result is '6002' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(6002) result is '"6002"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},"SourceChannel":${CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dcontext) result is 'from-internal' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(from-internal) result is '"from-internal"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(channel)})' (from 'CSV_QUOTE(${CDR(channel)})},"DestinationChannel":${CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(channel)' (from 'CDR(channel)})' len 12) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(channel) result is 'SIP/6001-00000000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6001-00000000) result is '"SIP/6001-00000000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dstchannel)})' (from 'CSV_QUOTE(${CDR(dstchannel)})},"LastAPP":${CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(dstchannel)' (from 'CDR(dstchannel)})' len 15) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(dstchannel) result is 'SIP/6003-00000003' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6003-00000003) result is '"SIP/6003-00000003"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastapp)})' (from 'CSV_QUOTE(${CDR(lastapp)})},"LastData":${CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 26) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(lastapp)' (from 'CDR(lastapp)})' len 12) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(lastapp) result is 'Dial' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(Dial) result is '"Dial"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(lastdata)})' (from 'CSV_QUOTE(${CDR(lastdata)})},"Start":${CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(lastdata)' (from 'CDR(lastdata)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(lastdata) result is 'SIP/6002,15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(SIP/6002,15) result is '"SIP/6002,15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},"Answer":${CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 24) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(start) result is '2014-03-21 17:24:15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},"End":${CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 25) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(answer) result is '2014-03-21 17:24:15' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:15) result is '"2014-03-21 17:24:15"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},"Duration":${CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 22) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(end) result is '2014-03-21 17:24:40' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(2014-03-21 17:24:40) result is '"2014-03-21 17:24:40"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})},"BillSec":${CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 29) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(duration,f) result is '0.025000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.025000) result is '"0.025000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},"Disposition":${CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(billsec,f) result is '0.025000' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(0.025000) result is '"0.025000"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(disposition)})' (from 'CSV_QUOTE(${CDR(disposition)})},"amaFlags":${CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(disposition)' (from 'CDR(disposition)})' len 16) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(disposition) result is 'ANSWERED' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(ANSWERED) result is '"ANSWERED"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(amaflags)})' (from 'CSV_QUOTE(${CDR(amaflags)})},"AccountCode":${CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(amaflags)' (from 'CDR(amaflags)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(amaflags) result is 'DOCUMENTATION' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(DOCUMENTATION) result is '"DOCUMENTATION"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(accountcode)})' (from 'CSV_QUOTE(${CDR(accountcode)})},"UniqueID":${CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 30) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(accountcode)' (from 'CDR(accountcode)})' len 16) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(accountcode) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(uniqueid)})' (from 'CSV_QUOTE(${CDR(uniqueid)})},"UserField":${CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 27) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(uniqueid)' (from 'CDR(uniqueid)})' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(uniqueid) result is '1395440637.0' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE(1395440637.0) result is '"1395440637.0"' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(userfield)})' (from 'CSV_QUOTE(${CDR(userfield)})},"Sequence":${CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 28) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(userfield)' (from 'CDR(userfield)})' len 14) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(userfield) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(sequence)' (from 'CDR(sequence)},"acustomfield1":${CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 13) [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(sequence) result is '5' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield1)})' (from 'CSV_QUOTE(${CDR(acustomfield1)})},"acustomfield2":${CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [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] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield1)' (from 'CDR(acustomfield1)})' len 18) [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 10.24.18.124:5060;rport=5060;received=10.24.18.124;branch=z9hG4bK6d9ee855 [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 2 [ 41]: Call-ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(acustomfield1) result is '' [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 3 [ 44]: From: ;tag=as28bb3d0e [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 4 [ 75]: To: "RustyONE" ;tag=m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 5 [ 13]: CSeq: 105 BYE [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CSV_QUOTE(${CDR(acustomfield2)})' (from 'CSV_QUOTE(${CDR(acustomfield2)})}} ' len 32) [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: --- (7 headers 0 lines) --- [Mar 21 17:24:40] DEBUG[10242] pbx.c: Evaluating 'CDR(acustomfield2)' (from 'CDR(acustomfield2)})' len 18) [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: = Looking for Call ID: Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ (Checking To) --From tag as28bb3d0e --To-tag m7bV6rpnYeBL36FCa35tzajXSvrIfo0m [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CDR(acustomfield2) result is '' [Mar 21 17:24:40] DEBUG[10242] pbx.c: Function CSV_QUOTE() result is '""' [Mar 21 17:24:40] DEBUG[10274][C-00000000] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #73 [Mar 21 17:24:40] DEBUG[10274][C-00000000] chan_sip.c: Stopping retransmission on 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' of Request 105: Match Found [Mar 21 17:24:40] DEBUG[10274] chan_sip.c: Destroying SIP dialog Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ [Mar 21 17:24:40] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog 'Q7uVWi.rH3vuBSs7Hk6mvOKUWjcqwGlQ' Method: ACK [Mar 21 17:24:40] DEBUG[10274] rtp_engine.c: Destroyed RTP instance '0x7f5d6802bae8' [Mar 21 17:24:41] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog '1096126b-1194f96a-fd46e4ad@10.24.18.180' [Mar 21 17:24:41] DEBUG[10274] chan_sip.c: Destroying SIP dialog 1096126b-1194f96a-fd46e4ad@10.24.18.180 [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] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' [Mar 21 17:24:48] DEBUG[10274] chan_sip.c: Destroying SIP dialog 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:48] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060' Method: BYE [Mar 21 17:24:48] DEBUG[10274] chan_sip.c: Updating call counter for outgoing call [Mar 21 17:24:48] DEBUG[10274] chan_sip.c: Call to peer '6002' removed from call limit 0 [Mar 21 17:24:48] DEBUG[10274] chan_sip.c: This call did not properly clean up call limits. Call ID 4b44c4fe2b1202b5613258035c3edc76@10.24.18.124:5060 [Mar 21 17:24:48] DEBUG[10240] devicestate.c: No provider found, checking channel drivers for SIP - 6002 [Mar 21 17:24:48] DEBUG[10240] chan_sip.c: Checking device state for peer 6002 [Mar 21 17:24:48] DEBUG[10240] devicestate.c: Changing state for SIP/6002 - state 1 (Not in use) [Mar 21 17:24:48] DEBUG[10274] rtp_engine.c: Destroyed RTP instance '0x7f5dc800b4f8' [Mar 21 17:24:57] DEBUG[10274] chan_sip.c: Auto destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' [Mar 21 17:24:57] DEBUG[10274] chan_sip.c: Destroying SIP dialog 21608363712124-1345167932089@10.24.18.166 [Mar 21 17:24:57] VERBOSE[10274] chan_sip.c: Really destroying SIP dialog '21608363712124-1345167932089@10.24.18.166' Method: REGISTER