[Jan 26 09:45:05] VERBOSE[24075] config.c: == Parsing '/etc/asterisk/logger.conf': [Jan 26 09:45:05] VERBOSE[24075] config.c: == Found [Jan 26 09:45:05] VERBOSE[24075] logger.c: Asterisk Queue Logger restarted [Jan 26 09:45:19] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> <-------------> [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> INVITE sip:10000009@ngn4.mydomian.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;rport Max-Forwards: 70 Contact: To: "10000009" From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 384 v=0 o=- 4 2 IN IP4 212.7.117.61 s=CounterPath eyeBeam 1.5 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 9 0 18 101 a=alt:1 3 : hXbTh93Y 8GffzGWT 192.168.1.10 55500 a=alt:2 2 : W0QV1/fO e9K1mkyg 192.168.56.1 55500 a=alt:3 1 : LPguLHpG ZhcmMXDv 212.7.117.61 55500 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 0 [ 48]: INVITE sip:10000009@ngn4.mydomian.com SIP/2.0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;rport [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 4 [ 49]: To: "10000009" [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 7 [ 14]: CSeq: 1 INVITE [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 10 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 11 [ 19]: Content-Length: 384 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 12 [ 0]: [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 4 2 IN IP4 212.7.117.61 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 212.7.117.61 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 5 [ 32]: m=audio 55500 RTP/AVP 9 0 18 101 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 6 [ 48]: a=alt:1 3 : hXbTh93Y 8GffzGWT 192.168.1.10 55500 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 7 [ 48]: a=alt:2 2 : W0QV1/fO e9K1mkyg 192.168.56.1 55500 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 8 [ 48]: a=alt:3 1 : LPguLHpG ZhcmMXDv 212.7.117.61 55500 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 9 [ 20]: a=fmtp:18 annexb=yes [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 11 [ 21]: a=rtpmap:18 G729/8000 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: --- (12 headers 14 lines) --- [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag [Jan 26 09:45:37] DEBUG[23737] acl.c: For destination '212.7.117.61', our source address is '208.211.92.75'. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 208.211.92.75:5060 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Allocating new SIP dialog for OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. - INVITE (No RTP) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 26 09:45:37] DEBUG[23737] netsock2.c: Splitting '192.168.1.10:19010' gives... [Jan 26 09:45:37] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '19010'. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Sending to 212.7.117.61:19010 (no NAT) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Initializing initreq for method INVITE - callid OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Using INVITE request as basis request - OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found peer 'dovid' for 'dovid' from 212.7.117.61:19010 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: <--- Reliably Transmitting (NAT) to 212.7.117.61:19010 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;received=212.7.117.61;rport=19010 From: "dovid";tag=4a482445 To: "10000009";tag=as2cabff25 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 1 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58bc190d" Content-Length: 0 <------------> [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #384 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Scheduling destruction of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' in 32000 ms (Method: INVITE) [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> ACK sip:10000009@ngn4.mydomian.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;rport To: "10000009";tag=as2cabff25 From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 1 ACK Content-Length: 0 <-------------> [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 0 [ 45]: ACK sip:10000009@ngn4.mydomian.com SIP/2.0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;rport [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 2 [ 64]: To: "10000009";tag=as2cabff25 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 3 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 4 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: --- (7 headers 0 lines) --- [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag as2cabff25 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #384 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Stopping retransmission on 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' of Response 1: Match Found [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> ACK sip:10000009@ngn4.mydomian.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;rport To: "10000009";tag=as2cabff25 From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 1 ACK Content-Length: 0 <-------------> [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 0 [ 45]: ACK sip:10000009@ngn4.mydomian.com SIP/2.0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-116aeb3220652d69-1---d8754z-;rport [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 2 [ 64]: To: "10000009";tag=as2cabff25 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 3 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 4 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: --- (7 headers 0 lines) --- [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag as2cabff25 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> INVITE sip:10000009@ngn4.mydomian.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-5342173b13497167-1---d8754z-;rport Max-Forwards: 70 Contact: To: "10000009" From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@ngn4.mydomian.com",response="194f41119d8be578e81724127d646a94",algorithm=MD5 Content-Length: 384 v=0 o=- 4 2 IN IP4 212.7.117.61 s=CounterPath eyeBeam 1.5 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 9 0 18 101 a=alt:1 3 : hXbTh93Y 8GffzGWT 192.168.1.10 55500 a=alt:2 2 : W0QV1/fO e9K1mkyg 192.168.56.1 55500 a=alt:3 1 : LPguLHpG ZhcmMXDv 212.7.117.61 55500 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 0 [ 48]: INVITE sip:10000009@ngn4.mydomian.com SIP/2.0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-5342173b13497167-1---d8754z-;rport [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 4 [ 49]: To: "10000009" [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 7 [ 14]: CSeq: 2 INVITE [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 10 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 11 [170]: Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@ngn4.mydomian.com",response="194f41119d8be578e81724127d646a94",algorithm=MD5 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 384 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Header 13 [ 0]: [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 4 2 IN IP4 212.7.117.61 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 212.7.117.61 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 5 [ 32]: m=audio 55500 RTP/AVP 9 0 18 101 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 6 [ 48]: a=alt:1 3 : hXbTh93Y 8GffzGWT 192.168.1.10 55500 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 7 [ 48]: a=alt:2 2 : W0QV1/fO e9K1mkyg 192.168.56.1 55500 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 8 [ 48]: a=alt:3 1 : LPguLHpG ZhcmMXDv 212.7.117.61 55500 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 9 [ 20]: a=fmtp:18 annexb=yes [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 11 [ 21]: a=rtpmap:18 G729/8000 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: --- (13 headers 14 lines) --- [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag [Jan 26 09:45:37] DEBUG[23737] netsock2.c: Splitting 'ngn4.mydomian.com' gives... [Jan 26 09:45:37] DEBUG[23737] netsock2.c: ...host 'ngn4.mydomian.com' and port '(null)'. [Jan 26 09:45:37] DEBUG[23737] netsock2.c: Splitting 'ngn4.mydomian.com' gives... [Jan 26 09:45:37] DEBUG[23737] netsock2.c: ...host 'ngn4.mydomian.com' and port '(null)'. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 26 09:45:37] DEBUG[23737] netsock2.c: Splitting '192.168.1.10:19010' gives... [Jan 26 09:45:37] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '19010'. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Sending to 212.7.117.61:19010 (NAT) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Initializing initreq for method INVITE - callid OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Using INVITE request as basis request - OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found peer 'dovid' for 'dovid' from 212.7.117.61:19010 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x993ce10' [Jan 26 09:45:37] DEBUG[23737] res_rtp_asterisk.c: Allocated port 17946 for RTP instance '0x993ce10' [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: RTP instance '0x993ce10' is setup and ready to go [Jan 26 09:45:37] DEBUG[23737] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x993ce10' [Jan 26 09:45:37] VERBOSE[23737] netsock2.c: == Using SIP RTP TOS bits 184 [Jan 26 09:45:37] VERBOSE[23737] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Setting NAT on RTP to On [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing session-level SDP o=- 4 2 IN IP4 212.7.117.61... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] netsock2.c: Splitting '212.7.117.61' gives... [Jan 26 09:45:37] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '(null)'. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 212.7.117.61... OK. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found RTP audio format 9 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Setting payload 9 based on m type on 0xb4508490 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found RTP audio format 0 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508490 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found RTP audio format 18 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Setting payload 18 based on m type on 0xb4508490 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found RTP audio format 101 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508490 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=alt:1 3 : hXbTh93Y 8GffzGWT 192.168.1.10 55500... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=alt:2 2 : W0QV1/fO e9K1mkyg 192.168.56.1 55500... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=alt:3 1 : LPguLHpG ZhcmMXDv 212.7.117.61 55500... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found audio description format G729 for ID 18 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508490 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Incorporating payload 9 on 0xb4508490 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Incorporating payload 18 on 0xb4508490 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508490 [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x1104 (ulaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 09:45:37] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x993ce10' [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 212.7.117.61:55500 [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508490 to 0x993cfbc [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Copying payload 9 from 0xb4508490 to 0x993cfbc [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Copying payload 18 from 0xb4508490 to 0x993cfbc [Jan 26 09:45:37] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508490 to 0x993cfbc [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Checking SIP call limits for device dovid [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Updating call counter for incoming call [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: Looking for 10000009 in dovid (domain ngn4.mydomian.com) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: This channel will not be able to handle video. [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: build_route: Contact hop: [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: list_route: hop: [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: SIP/dovid-00000030: New call is still down.... Trying... [Jan 26 09:45:37] VERBOSE[23737] chan_sip.c: <--- Transmitting (NAT) to 212.7.117.61:19010 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-5342173b13497167-1---d8754z-;received=212.7.117.61;rport=19010 From: "dovid";tag=4a482445 To: "10000009" Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 26 09:45:37] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:37] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - dovid [Jan 26 09:45:37] DEBUG[23716] chan_sip.c: Checking device state for peer dovid [Jan 26 09:45:37] DEBUG[23716] devicestate.c: Changing state for SIP/dovid - state 1 (Not in use) [Jan 26 09:45:37] DEBUG[23716] devicestate.c: device 'SIP/dovid' state '1' [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/dovid-00000030 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: dovid CallerIDName: dovid AccountCode: Exten: 10000009 Context: dovid Uniqueid: 1296053137.48 [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: SIPURI Value: sip:dovid@212.7.117.61:19010 Uniqueid: 1296053137.48 [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: SIPDOMAIN Value: ngn4.mydomian.com Uniqueid: 1296053137.48 [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: SIPCALLID Value: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. Uniqueid: 1296053137.48 [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/dovid-00000030 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: dovid CallerIDName: dovid Uniqueid: 1296053137.48 [Jan 26 09:45:37] DEBUG[24077] pbx.c: Launching 'Answer' [Jan 26 09:45:37] VERBOSE[24077] pbx.c: -- Executing [10000009@dovid:1] Answer("SIP/dovid-00000030", "") in new stack [Jan 26 09:45:37] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - dovid [Jan 26 09:45:37] DEBUG[23716] chan_sip.c: Checking device state for peer dovid [Jan 26 09:45:37] DEBUG[23716] devicestate.c: Changing state for SIP/dovid - state 1 (Not in use) [Jan 26 09:45:37] DEBUG[23716] devicestate.c: device 'SIP/dovid' state '1' [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: SIP answering channel: SIP/dovid-00000030 [Jan 26 09:45:37] DEBUG[24077] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: Setting framing from config on incoming call [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 26 09:45:37] VERBOSE[24077] chan_sip.c: Audio is at 5060 [Jan 26 09:45:37] VERBOSE[24077] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:37] VERBOSE[24077] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:37] VERBOSE[24077] chan_sip.c: <--- Reliably Transmitting (NAT) to 212.7.117.61:19010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-5342173b13497167-1---d8754z-;received=212.7.117.61;rport=19010 From: "dovid";tag=4a482445 To: "10000009";tag=as37f9acee Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 1047290220 1047290220 IN IP4 208.211.92.75 s=Asterisk PBX 1.8.2.2 c=IN IP4 208.211.92.75 t=0 0 m=audio 17946 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #387 [Jan 26 09:45:37] DEBUG[24077] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/dovid-00000030 Context: dovid Extension: 10000009 Priority: 1 Application: Answer AppData: Uniqueid: 1296053137.48 [Jan 26 09:45:37] DEBUG[23750] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/dovid-00000030 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: dovid CallerIDName: dovid Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> ACK sip:10000009@208.211.92.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-6308555a84692642-1---d8754z-;rport Max-Forwards: 70 Contact: To: "10000009";tag=as37f9acee From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 2 ACK User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@ngn4.mydomian.com",response="194f41119d8be578e81724127d646a94",algorithm=MD5 Content-Length: 0 <-------------> [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 0 [ 43]: ACK sip:10000009@208.211.92.75:5060 SIP/2.0 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-6308555a84692642-1---d8754z-;rport [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 4 [ 64]: To: "10000009";tag=as37f9acee [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 8 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 9 [170]: Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@ngn4.mydomian.com",response="194f41119d8be578e81724127d646a94",algorithm=MD5 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: --- (11 headers 0 lines) --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag as37f9acee [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #387 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Stopping retransmission on 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' of Response 2: Match Found [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> ACK sip:10000009@208.211.92.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-6308555a84692642-1---d8754z-;rport Max-Forwards: 70 Contact: To: "10000009";tag=as37f9acee From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 2 ACK User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@ngn4.mydomian.com",response="194f41119d8be578e81724127d646a94",algorithm=MD5 Content-Length: 0 <-------------> [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 0 [ 43]: ACK sip:10000009@208.211.92.75:5060 SIP/2.0 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-6308555a84692642-1---d8754z-;rport [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 4 [ 64]: To: "10000009";tag=as37f9acee [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 8 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 9 [170]: Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@ngn4.mydomian.com",response="194f41119d8be578e81724127d646a94",algorithm=MD5 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: --- (11 headers 0 lines) --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag as37f9acee [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 26 09:45:38] DEBUG[24077] pbx.c: Result of 'EXTEN' is '10000009' [Jan 26 09:45:38] DEBUG[24077] pbx.c: Launching 'Dial' [Jan 26 09:45:38] VERBOSE[24077] pbx.c: -- Executing [10000009@dovid:2] Dial("SIP/dovid-00000030", "SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)F(db_test^1^1)") in new stack [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Allocating new SIP dialog for 359cb2057ef8db7d5e7d375b43521099@208.211.92.75:0 - INVITE (No RTP) [Jan 26 09:45:38] DEBUG[24077] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xd10dec8' [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: Allocated port 16866 for RTP instance '0xd10dec8' [Jan 26 09:45:38] DEBUG[24077] rtp_engine.c: RTP instance '0xd10dec8' is setup and ready to go [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xd10dec8' [Jan 26 09:45:38] VERBOSE[24077] netsock2.c: == Using SIP RTP TOS bits 184 [Jan 26 09:45:38] VERBOSE[24077] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Setting NAT on RTP to Off [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 26 09:45:38] DEBUG[24077] acl.c: For destination '69.167.68.130', our source address is '208.211.92.75'. [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 208.211.92.75:5060 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: This channel will not be able to handle video. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable DIALEDTIME. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable ANSWEREDTIME. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable DIALEDPEERNAME. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable DIALEDPEERNUMBER. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable DIALSTATUS. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable SIPCALLID. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable SIPDOMAIN. [Jan 26 09:45:38] DEBUG[24077] channel.c: Not copying variable SIPURI. [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Outgoing Call for 10000009 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Updating call counter for outgoing call [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 26 09:45:38] VERBOSE[24077] chan_sip.c: Audio is at 5060 [Jan 26 09:45:38] VERBOSE[24077] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:38] VERBOSE[24077] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 26 09:45:38] VERBOSE[24077] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Initializing initreq for method INVITE - callid 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 0 [ 41]: INVITE sip:10000009@69.167.68.130 SIP/2.0 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f6352f2 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 3 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 4 [ 32]: To: [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 5 [ 39]: Contact: [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 6 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 9 [ 35]: Date: Wed, 26 Jan 2011 14:45:38 GMT [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 26 09:45:38] VERBOSE[24077] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060: INVITE sip:10000009@69.167.68.130 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f6352f2 Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.2 Date: Wed, 26 Jan 2011 14:45:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1043253688 1043253688 IN IP4 208.211.92.75 s=Asterisk PBX 1.8.2.2 c=IN IP4 208.211.92.75 t=0 0 m=audio 16866 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #389 [Jan 26 09:45:38] DEBUG[24077] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:38] VERBOSE[24077] app_dial.c: -- Called 10000009@fpp [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/dovid-00000030 Context: dovid Extension: 10000009 Priority: 2 Application: Dial AppData: SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)F(db_test^1^1) Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALSTATUS Value: Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALEDPEERNUMBER Value: Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALEDPEERNAME Value: Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ANSWEREDTIME Value: Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALEDTIME Value: Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: Newchannel Privilege: call,all Channel: SIP/fpp-00000031 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: from-sip Uniqueid: 1296053138.49 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: SIPCALLID Value: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 Uniqueid: 1296053138.49 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: DIALEDPEERNUMBER Value: 10000009@fpp Uniqueid: 1296053138.49 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/dovid-00000030 Destination: SIP/fpp-00000031 CallerIDNum: dovid CallerIDName: dovid UniqueID: 1296053137.48 DestUniqueID: 1296053138.49 Dialstring: 10000009@fpp [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: NewCallerid Privilege: call,all Channel: SIP/fpp-00000031 CallerIDNum: 10000009 CallerIDName: Uniqueid: 1296053138.49 CID-CallingPres: 0 (Presentation Allowed, Not Screened) [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f6352f2 From: "dovid" ;tag=as405d8e89 To: ;tag=8a7940c898c7113a1fb8a4a76e6676f0.9560 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4033b0000001419929e687bc4e7f383801bc682bdbf822" Server: PBX_MANAGER Content-Length: 0 Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27132 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.130 via_cnt==1" <-------------> [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f6352f2 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 3 [ 74]: To: ;tag=8a7940c898c7113a1fb8a4a76e6676f0.9560 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4033b0000001419929e687bc4e7f383801bc682bdbf822" [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 9 [188]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27132 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.130 via_cnt==1" [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag 8a7940c898c7113a1fb8a4a76e6676f0.9560 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Acked pending invite 102 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #389 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Stopping retransmission on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' of Request 102: Match Found [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: SIP response 407 to standard invite [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060: ACK sip:10000009@69.167.68.130 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2f6352f2 Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=8a7940c898c7113a1fb8a4a76e6676f0.9560 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: Audio is at 5060 [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060: INVITE sip:10000009@69.167.68.130 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.2.2 Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.130", nonce="4d4033b0000001419929e687bc4e7f383801bc682bdbf822", response="552a5e5521e858e0324220c5508f8567" Date: Wed, 26 Jan 2011 14:45:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1043253688 1043253689 IN IP4 208.211.92.75 s=Asterisk PBX 1.8.2.2 c=IN IP4 208.211.92.75 t=0 0 m=audio 16866 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #391 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 407 Proxy Authentication Required Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 From: "dovid" ;tag=as405d8e89 To: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 103 INVITE Server: PBX_MANAGER Content-Length: 0 Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.134:5060;transport=udp via_cnt==1" <-------------> [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 3 [ 32]: To: [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 8 [207]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.130 out_uri=sip:10000009@69.167.68.134:5060;transport=udp via_cnt==1" [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #391 - INVITE (got response) [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Request 103: Found [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: SIP response 100 to standard invite [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 100 Giving a try Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 Record-Route: From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 103 INVITE Server: PBX_MANAGER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <-------------> [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 103 INVITE [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 26 09:45:38] VERBOSE[23737] chan_sip.c: --- (12 headers 0 lines) --- [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Request 103: Found [Jan 26 09:45:38] DEBUG[23737] chan_sip.c: SIP response 180 to standard invite [Jan 26 09:45:38] VERBOSE[24077] app_dial.c: -- SIP/fpp-00000031 is ringing [Jan 26 09:45:38] DEBUG[24077] channel.c: Driver for channel 'SIP/dovid-00000030' does not support indication 3, emulating it [Jan 26 09:45:38] DEBUG[24077] channel.c: Set channel SIP/dovid-00000030 to write format slin [Jan 26 09:45:38] DEBUG[24077] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 09:45:38] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - fpp [Jan 26 09:45:38] DEBUG[23716] chan_sip.c: Checking device state for peer fpp [Jan 26 09:45:38] DEBUG[23716] devicestate.c: Changing state for SIP/fpp - state 1 (Not in use) [Jan 26 09:45:38] DEBUG[23716] devicestate.c: device 'SIP/fpp' state '1' [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/fpp-00000031 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 10000009 CallerIDName: Uniqueid: 1296053138.49 [Jan 26 09:45:38] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 180 Ringing Uniqueid: 1296053137.48 [Jan 26 09:45:38] DEBUG[24077] channel.c: Generator got voice, switching to phase locked mode [Jan 26 09:45:38] DEBUG[24077] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:38] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:39] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: No remote address on RTP instance '0xd10dec8' so dropping frame [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 Record-Route: From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 103 INVITE Server: PBX_MANAGER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 2087990304 2087990304 IN IP4 69.167.68.134 s=PBX_MANAGER c=IN IP4 69.167.68.134 t=0 0 m=audio 12022 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=direction:active <-------------> [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK106c3514 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 103 INVITE [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 300 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Header 13 [ 0]: [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 1 [ 49]: o=root 2087990304 2087990304 IN IP4 69.167.68.134 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.134 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 5 [ 29]: m=audio 12022 RTP/AVP 0 8 101 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-16 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 10 [ 25]: a=silenceSupp:off - - - - [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 12 [ 10]: a=sendrecv [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Body 13 [ 18]: a=direction:active [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: --- (13 headers 14 lines) --- [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Acked pending invite 103 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Stopping retransmission on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' of Request 103: Match Found [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: SIP response 200 to standard invite [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 2087990304 2087990304 IN IP4 69.167.68.134... UNSUPPORTED. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED. [Jan 26 09:45:40] DEBUG[23737] netsock2.c: Splitting '69.167.68.134' gives... [Jan 26 09:45:40] DEBUG[23737] netsock2.c: ...host '69.167.68.134' and port '(null)'. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.134... OK. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Found RTP audio format 0 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100 [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Found RTP audio format 8 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Setting payload 8 based on m type on 0xb4508100 [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Found RTP audio format 101 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100 [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED. [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Incorporating payload 8 on 0xb4508100 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100 [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 09:45:40] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd10dec8' [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.134:12022 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Copying payload 8 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:40] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: build_route: Record-Route hop: [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: list_route: hop: [Jan 26 09:45:40] DEBUG[23737] netsock2.c: Splitting '69.167.68.134:5060' gives... [Jan 26 09:45:40] DEBUG[23737] netsock2.c: ...host '69.167.68.134' and port '5060'. [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:40] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:40] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:40] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060: ACK sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK7a2f34db Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:40] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:40] VERBOSE[24077] app_dial.c: -- SIP/fpp-00000031 answered SIP/dovid-00000030 [Jan 26 09:45:40] DEBUG[24077] channel.c: Set channel SIP/dovid-00000030 to write format ulaw [Jan 26 09:45:40] DEBUG[24077] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 09:45:40] DEBUG[24077] app_stack.c: Channel SIP/fpp-00000031 has no datastore, so we're allocating one. [Jan 26 09:45:40] DEBUG[24077] app_stack.c: Setting 'ARG1' to 's' [Jan 26 09:45:40] DEBUG[24077] app_stack.c: Setting 'ARG2' to '1' [Jan 26 09:45:40] DEBUG[24077] pbx.c: Launching 'AGI' [Jan 26 09:45:40] VERBOSE[24077] pbx.c: -- Executing [s@do_dtmf_cc-take-call:1] AGI("SIP/fpp-00000031", "agi://127.0.0.1:4579/update_call_status?status=60") in new stack [Jan 26 09:45:40] DEBUG[24077] res_agi.c: Wow, connected! [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 200 OK Uniqueid: 1296053137.48 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: Newstate Privilege: call,all Channel: SIP/fpp-00000031 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 10000009 CallerIDName: Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1296053137.48 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALEDPEERNAME Value: SIP/fpp-00000031 Uniqueid: 1296053137.48 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALEDPEERNUMBER Value: 10000009@fpp Uniqueid: 1296053137.48 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: LOCAL(ARG1) Value: s Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: LOCAL(ARG2) Value: 1 Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: LOCAL(ARGC) Value: 2 Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/fpp-00000031 Context: do_dtmf_cc-take-call Extension: s Priority: 1 Application: AGI AppData: agi://127.0.0.1:4579/update_call_status?status=60 Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - fpp [Jan 26 09:45:40] DEBUG[23716] chan_sip.c: Checking device state for peer fpp [Jan 26 09:45:40] DEBUG[23716] devicestate.c: Changing state for SIP/fpp - state 1 (Not in use) [Jan 26 09:45:40] DEBUG[23716] devicestate.c: device 'SIP/fpp' state '1' [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'our_start' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1809669776 Command: GET VARIABLE our_start [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1809669776 Command: GET VARIABLE our_start ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'uuid' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1268060586 Command: GET VARIABLE uuid [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1268060586 Command: GET VARIABLE uuid ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'recording' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 2077541652 Command: GET VARIABLE recording [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 2077541652 Command: GET VARIABLE recording ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'rec_file' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 582924010 Command: GET VARIABLE rec_file [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 582924010 Command: GET VARIABLE rec_file ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'pass' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 844439104 Command: GET VARIABLE pass [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 844439104 Command: GET VARIABLE pass ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'lega' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 701975677 Command: GET VARIABLE lega [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 701975677 Command: GET VARIABLE lega ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'legb' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 142904543 Command: GET VARIABLE legb [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 142904543 Command: GET VARIABLE legb ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'cida' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1294382474 Command: GET VARIABLE cida [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1294382474 Command: GET VARIABLE cida ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'cidb' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1147800838 Command: GET VARIABLE cidb [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1147800838 Command: GET VARIABLE cidb ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'send_dtmf' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1127945249 Command: GET VARIABLE send_dtmf [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1127945249 Command: GET VARIABLE send_dtmf ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'dtmf' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 64356211 Command: GET VARIABLE dtmf [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 64356211 Command: GET VARIABLE dtmf ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'wava1' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1858578160 Command: GET VARIABLE wava1 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1858578160 Command: GET VARIABLE wava1 ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'play_wava2' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1137619262 Command: GET VARIABLE play_wava2 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1137619262 Command: GET VARIABLE play_wava2 ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'wava2' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1799508680 Command: GET VARIABLE wava2 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1799508680 Command: GET VARIABLE wava2 ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'play_wavb' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 239299984 Command: GET VARIABLE play_wavb [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 239299984 Command: GET VARIABLE play_wavb ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'wavb' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1678565159 Command: GET VARIABLE wavb [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1678565159 Command: GET VARIABLE wavb ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'play_wava3' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 203888051 Command: GET VARIABLE play_wava3 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 203888051 Command: GET VARIABLE play_wava3 ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'wava3' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1916531652 Command: GET VARIABLE wava3 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1916531652 Command: GET VARIABLE wava3 ResultCode: 200 Result: Success [Jan 26 09:45:40] DEBUG[24077] pbx.c: Result of 'timeout' is NULL [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 515011050 Command: GET VARIABLE timeout [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 515011050 Command: GET VARIABLE timeout ResultCode: 200 Result: Success [Jan 26 09:45:40] VERBOSE[24077] res_agi.c: -- AGI Script agi://127.0.0.1:4579/update_call_status?status=60 completed, returning 0 [Jan 26 09:45:40] DEBUG[24077] pbx.c: Launching 'SendDTMF' [Jan 26 09:45:40] VERBOSE[24077] pbx.c: -- Executing [s@do_dtmf_cc-take-call:2] SendDTMF("SIP/fpp-00000031", "123456") in new stack [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: AGISTATUS Value: SUCCESS Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/fpp-00000031 Context: do_dtmf_cc-take-call Extension: s Priority: 2 Application: SendDTMF AppData: 123456 Uniqueid: 1296053138.49 [Jan 26 09:45:40] DEBUG[24077] res_rtp_asterisk.c: Adjusting final end duration from 640 to 800 [Jan 26 09:45:41] DEBUG[23754] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 26 09:45:41] DEBUG[23750] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 212.7.117.61:55501 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 40835 SequenceNumberCycles: 0 IAJitter: 45 LastSR: 0.0000000000 DLSR: 0.0000(sec) [Jan 26 09:45:42] DEBUG[24077] pbx.c: Launching 'AGI' [Jan 26 09:45:42] VERBOSE[24077] pbx.c: -- Executing [s@do_dtmf_cc-take-call:3] AGI("SIP/fpp-00000031", "agi://127.0.0.1:4579/update_call_status?status=70") in new stack [Jan 26 09:45:42] DEBUG[24077] res_agi.c: Wow, connected! [Jan 26 09:45:42] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/fpp-00000031 Context: do_dtmf_cc-take-call Extension: s Priority: 3 Application: AGI AppData: agi://127.0.0.1:4579/update_call_status?status=70 Uniqueid: 1296053138.49 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1538441437 Command: GET VARIABLE our_start [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'our_start' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1538441437 Command: GET VARIABLE our_start ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1674813443 Command: GET VARIABLE uuid [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'uuid' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1674813443 Command: GET VARIABLE uuid ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'recording' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1919003810 Command: GET VARIABLE recording [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1919003810 Command: GET VARIABLE recording ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'rec_file' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1418695411 Command: GET VARIABLE rec_file [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1418695411 Command: GET VARIABLE rec_file ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'pass' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1441129553 Command: GET VARIABLE pass [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1441129553 Command: GET VARIABLE pass ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'lega' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1586105084 Command: GET VARIABLE lega [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1586105084 Command: GET VARIABLE lega ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'legb' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1587657372 Command: GET VARIABLE legb [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1587657372 Command: GET VARIABLE legb ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'cida' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1840335779 Command: GET VARIABLE cida [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1840335779 Command: GET VARIABLE cida ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'cidb' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1397883012 Command: GET VARIABLE cidb [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1397883012 Command: GET VARIABLE cidb ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'send_dtmf' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 572363098 Command: GET VARIABLE send_dtmf [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 572363098 Command: GET VARIABLE send_dtmf ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 212.7.117.61:55501 OurSSRC: 613198591 SentNTP: 1296053143.0250519552 SentRTP: 18720 SentPackets: 117 SentOctets: 18720 ReportBlock: FractionLost: 4 CumulativeLoss: 4 IAJitter: 0.0045 TheirLastSR: 2988038094 DLSR: 1.9790 (sec) [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'dtmf' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1916460602 Command: GET VARIABLE dtmf [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1916460602 Command: GET VARIABLE dtmf ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'wava1' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1484901191 Command: GET VARIABLE wava1 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1484901191 Command: GET VARIABLE wava1 ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'play_wava2' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 598036805 Command: GET VARIABLE play_wava2 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 598036805 Command: GET VARIABLE play_wava2 ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'wava2' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1003052022 Command: GET VARIABLE wava2 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1003052022 Command: GET VARIABLE wava2 ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'play_wavb' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1272616026 Command: GET VARIABLE play_wavb [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1272616026 Command: GET VARIABLE play_wavb ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'wavb' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 421536000 Command: GET VARIABLE wavb [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 421536000 Command: GET VARIABLE wavb ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'play_wava3' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 96492157 Command: GET VARIABLE play_wava3 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 96492157 Command: GET VARIABLE play_wava3 ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'wava3' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 408597976 Command: GET VARIABLE wava3 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 408597976 Command: GET VARIABLE wava3 ResultCode: 200 Result: Success [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'timeout' is NULL [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 2052163876 Command: GET VARIABLE timeout [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 2052163876 Command: GET VARIABLE timeout ResultCode: 200 Result: Success [Jan 26 09:45:43] VERBOSE[24077] res_agi.c: -- AGI Script agi://127.0.0.1:4579/update_call_status?status=70 completed, returning 0 [Jan 26 09:45:43] DEBUG[24077] pbx.c: Result of 'EPOCH' is '1296053143' [Jan 26 09:45:43] DEBUG[24077] pbx.c: Launching 'Set' [Jan 26 09:45:43] VERBOSE[24077] pbx.c: -- Executing [s@do_dtmf_cc-take-call:4] Set("SIP/fpp-00000031", "wavb_start=1296053143") in new stack [Jan 26 09:45:43] DEBUG[24077] pbx.c: Launching 'BackGround' [Jan 26 09:45:43] VERBOSE[24077] pbx.c: -- Executing [s@do_dtmf_cc-take-call:5] BackGround("SIP/fpp-00000031", "/etc/cb/wav/incoming_cb_call") in new stack [Jan 26 09:45:43] DEBUG[24077] channel.c: Set channel SIP/fpp-00000031 to write format gsm [Jan 26 09:45:43] DEBUG[24077] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Jan 26 09:45:43] DEBUG[24077] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Jan 26 09:45:43] DEBUG[24077] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 09:45:43] VERBOSE[24077] file.c: -- Playing '/etc/cb/wav/incoming_cb_call.gsm' (language 'en') [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: AGISTATUS Value: SUCCESS Uniqueid: 1296053138.49 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/fpp-00000031 Context: do_dtmf_cc-take-call Extension: s Priority: 4 Application: Set AppData: wavb_start=1296053143 Uniqueid: 1296053138.49 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: wavb_start Value: 1296053143 Uniqueid: 1296053138.49 [Jan 26 09:45:43] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/fpp-00000031 Context: do_dtmf_cc-take-call Extension: s Priority: 5 Application: BackGround AppData: /etc/cb/wav/incoming_cb_call Uniqueid: 1296053138.49 [Jan 26 09:45:44] DEBUG[23754] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 26 09:45:44] DEBUG[23750] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 212.7.117.61:55501 PT: 200(Sender Report) ReceptionReports: 1 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 40835 SequenceNumberCycles: 0 IAJitter: 45 LastSR: 45591.2147483648 DLSR: 0.8860(sec) RTT: 174(sec) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: Sending dtmf: 49 (1), at 69.167.68.134:12022 [Jan 26 09:45:45] DTMF[24077] channel.c: DTMF begin '1' received on SIP/fpp-00000031 [Jan 26 09:45:45] DTMF[24077] channel.c: DTMF begin ignored '1' on SIP/fpp-00000031 [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: DTMF Privilege: dtmf,all Channel: SIP/fpp-00000031 Uniqueid: 1296053138.49 Digit: 1 Direction: Received Begin: Yes End: No [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: Sending dtmf: 49 (1), at 69.167.68.134:12022 [Jan 26 09:45:45] DTMF[24077] channel.c: DTMF end '1' received on SIP/fpp-00000031, duration 160 ms [Jan 26 09:45:45] DTMF[24077] channel.c: DTMF end passthrough '1' on SIP/fpp-00000031 [Jan 26 09:45:45] DEBUG[24077] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 09:45:45] DEBUG[24077] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 09:45:45] DEBUG[24077] channel.c: Set channel SIP/fpp-00000031 to write format ulaw [Jan 26 09:45:45] DEBUG[24077] pbx.c: Launching 'AGI' [Jan 26 09:45:45] VERBOSE[24077] pbx.c: -- Executing [1@do_dtmf_cc-take-call:1] AGI("SIP/fpp-00000031", "agi://127.0.0.1:4579/update_call_status?status=80") in new stack [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: DTMF Privilege: dtmf,all Channel: SIP/fpp-00000031 Uniqueid: 1296053138.49 Digit: 1 Direction: Received Begin: No End: Yes [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: BACKGROUNDSTATUS Value: SUCCESS Uniqueid: 1296053138.49 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/fpp-00000031 Context: do_dtmf_cc-take-call Extension: 1 Priority: 1 Application: AGI AppData: agi://127.0.0.1:4579/update_call_status?status=80 Uniqueid: 1296053138.49 [Jan 26 09:45:45] DEBUG[24077] res_agi.c: Wow, connected! [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'our_start' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1072452694 Command: GET VARIABLE our_start [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1072452694 Command: GET VARIABLE our_start ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 2000336966 Command: GET VARIABLE uuid [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'uuid' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 2000336966 Command: GET VARIABLE uuid ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'recording' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 553689328 Command: GET VARIABLE recording [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 553689328 Command: GET VARIABLE recording ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'rec_file' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 460220685 Command: GET VARIABLE rec_file [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 460220685 Command: GET VARIABLE rec_file ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'pass' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 929124463 Command: GET VARIABLE pass [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 929124463 Command: GET VARIABLE pass ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'lega' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 879514771 Command: GET VARIABLE lega [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 879514771 Command: GET VARIABLE lega ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'legb' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1735432847 Command: GET VARIABLE legb [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1735432847 Command: GET VARIABLE legb ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'cida' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 694793597 Command: GET VARIABLE cida [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 694793597 Command: GET VARIABLE cida ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'cidb' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1113180463 Command: GET VARIABLE cidb [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1113180463 Command: GET VARIABLE cidb ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'send_dtmf' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1488666479 Command: GET VARIABLE send_dtmf [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1488666479 Command: GET VARIABLE send_dtmf ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'dtmf' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1669621710 Command: GET VARIABLE dtmf [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1669621710 Command: GET VARIABLE dtmf ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'wava1' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1583434624 Command: GET VARIABLE wava1 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1583434624 Command: GET VARIABLE wava1 ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'play_wava2' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1465308431 Command: GET VARIABLE play_wava2 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1465308431 Command: GET VARIABLE play_wava2 ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'wava2' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1064480312 Command: GET VARIABLE wava2 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1064480312 Command: GET VARIABLE wava2 ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'play_wavb' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1884253455 Command: GET VARIABLE play_wavb [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1884253455 Command: GET VARIABLE play_wavb ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'wavb' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 793171950 Command: GET VARIABLE wavb [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 793171950 Command: GET VARIABLE wavb ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'play_wava3' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1713758295 Command: GET VARIABLE play_wava3 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1713758295 Command: GET VARIABLE play_wava3 ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'wava3' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 183143135 Command: GET VARIABLE wava3 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 183143135 Command: GET VARIABLE wava3 ResultCode: 200 Result: Success [Jan 26 09:45:45] DEBUG[24077] pbx.c: Result of 'timeout' is NULL [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/fpp-00000031 CommandId: 1933940870 Command: GET VARIABLE timeout [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: AGIExec Privilege: agi,all SubEvent: End Channel: SIP/fpp-00000031 CommandId: 1933940870 Command: GET VARIABLE timeout ResultCode: 200 Result: Success [Jan 26 09:45:45] VERBOSE[24077] res_agi.c: -- AGI Script agi://127.0.0.1:4579/update_call_status?status=80 completed, returning 0 [Jan 26 09:45:45] VERBOSE[24077] pbx.c: -- Auto fallthrough, channel 'SIP/fpp-00000031' status is 'UNKNOWN' [Jan 26 09:45:45] DEBUG[24077] app_dial.c: Gosub exited with status 0 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: AGISTATUS Value: SUCCESS Uniqueid: 1296053138.49 [Jan 26 09:45:45] DEBUG[24077] features.c: bridge answer set, chan answer set [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 26 09:45:45] DEBUG[24077] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 26 09:45:45] VERBOSE[24077] rtp_engine.c: -- Remotely bridging SIP/dovid-00000030 and SIP/fpp-00000031 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Sending reinvite on SIP 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' - It's audio soon redirected to IP 69.167.68.134:12022 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Strict routing enforced for session OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[24077] netsock2.c: Splitting '212.7.117.61:19010' gives... [Jan 26 09:45:45] DEBUG[24077] netsock2.c: ...host '212.7.117.61' and port '19010'. [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: set_destination: set destination to 212.7.117.61:19010 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Audio is at 5060 [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Initializing already initialized SIP dialog OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (presumably reinvite) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 0 [ 43]: INVITE sip:dovid@212.7.117.61:19010 SIP/2.0 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4dd188dc;rport [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 3 [ 66]: From: "10000009";tag=as37f9acee [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 4 [ 56]: To: "dovid";tag=4a482445 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 5 [ 42]: Contact: [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Reliably Transmitting (NAT) to 212.7.117.61:19010: INVITE sip:dovid@212.7.117.61:19010 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4dd188dc;rport Max-Forwards: 70 From: "10000009";tag=as37f9acee To: "dovid";tag=4a482445 Contact: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.2 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: 265 v=0 o=root 1047290220 1047290221 IN IP4 69.167.68.134 s=Asterisk PBX 1.8.2.2 c=IN IP4 69.167.68.134 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #393 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Sending reinvite on SIP '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' - It's audio soon redirected to IP 212.7.117.61:55500 [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[24077] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[24077] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Audio is at 5060 [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Initializing already initialized SIP dialog 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (presumably reinvite) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 0 [ 46]: INVITE sip:10000009@69.167.68.134:5060 SIP/2.0 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK42bde4fa [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 2 [ 48]: Route: [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 4 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 5 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 6 [ 39]: Contact: [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 7 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 8 [ 16]: CSeq: 104 INVITE [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 9 [ 32]: User-Agent: Asterisk PBX 1.8.2.2 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060: INVITE sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK42bde4fa Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.2.2 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: 263 v=0 o=root 1043253688 1043253690 IN IP4 212.7.117.61 s=Asterisk PBX 1.8.2.2 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #394 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: BRIDGEPEER Value: SIP/fpp-00000031 Uniqueid: 1296053137.48 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: BRIDGEPEER Value: SIP/dovid-00000030 Uniqueid: 1296053138.49 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: NewAccountCode Privilege: call,all Channel: SIP/fpp-00000031 Uniqueid: 1296053138.49 AccountCode: OldAccountCode: [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/dovid-00000030 Channel2: SIP/fpp-00000031 Uniqueid1: 1296053137.48 Uniqueid2: 1296053138.49 CallerID1: dovid CallerID2: 10000009 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: BRIDGEPEER Value: SIP/fpp-00000031 Uniqueid: 1296053137.48 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: BRIDGEPVTCALLID Value: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 Uniqueid: 1296053137.48 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: BRIDGEPEER Value: SIP/dovid-00000030 Uniqueid: 1296053138.49 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: BRIDGEPVTCALLID Value: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. Uniqueid: 1296053138.49 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK42bde4fa From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 104 INVITE Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4033b700000279be5cf413829850d96ee5d14e491568ef" Server: PBX_MANAGER Content-Length: 0 Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK42bde4fa [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4033b700000279be5cf413829850d96ee5d14e491568ef" [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27128 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Acked pending invite 104 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #394 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Stopping retransmission on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' of Request 104: Match Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 407 to RE-invite on outgoing call 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060: ACK sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK42bde4fa Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Audio is at 5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060: INVITE sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK182b1b2a Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.2.2 Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.134:5060", nonce="4d4033b700000279be5cf413829850d96ee5d14e491568ef", response="3dc0e4e4f4dc106fdc79117faf9f88dc" Date: Wed, 26 Jan 2011 14:45:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 1043253688 1043253691 IN IP4 212.7.117.61 s=Asterisk PBX 1.8.2.2 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #395 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 407 Proxy Authentication Required Uniqueid: 1296053137.48 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK182b1b2a From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 105 INVITE Server: PBX_MANAGER Content-Length: 0 Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27134 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK182b1b2a [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 105 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27134 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #395 - INVITE (got response) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Request 105: Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 100 to RE-invite on outgoing call 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 100 Giving a try Uniqueid: 1296053137.48 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK182b1b2a Record-Route: From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 105 INVITE Server: PBX_MANAGER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 276 v=0 o=root 2087990304 2087990305 IN IP4 69.167.68.134 s=PBX_MANAGER c=IN IP4 69.167.68.134 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=direction:active <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK182b1b2a [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 105 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 276 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 13 [ 0]: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 1 [ 49]: o=root 2087990304 2087990305 IN IP4 69.167.68.134 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.134 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 12022 RTP/AVP 0 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=sendrecv [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 12 [ 18]: a=direction:active [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (13 headers 13 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Acked pending invite 105 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Stopping retransmission on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' of Request 105: Match Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 2087990304 2087990305 IN IP4 69.167.68.134... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.134' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.134' and port '(null)'. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.134... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found RTP audio format 0 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found RTP audio format 101 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 09:45:45] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd10dec8' [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.134:12022 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.134:5060' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.134' and port '5060'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060: ACK sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK5d1400a8 Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/fpp-00000031' changed end address to 69.167.68.134:12022 (format ulaw) [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/fpp-00000031' changed end vaddress to (null) (format ulaw) [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/fpp-00000031' changed end taddress to (null) (format ulaw) [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/fpp-00000031' was 69.167.68.134:12022/(format unknown) [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/fpp-00000031' was (null)/(format unknown) [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/fpp-00000031' was (null)/(format unknown) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Deferring reinvite on SIP 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' - It's audio will be redirected to IP 69.167.68.134:12022 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 200 OK Uniqueid: 1296053137.48 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4dd188dc;rport=5060 Contact: To: "dovid";tag=4a482445 From: "10000009";tag=as37f9acee Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 4 3 IN IP4 212.7.117.61 s=CounterPath eyeBeam 1.5 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK4dd188dc;rport=5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 39]: Contact: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 56]: To: "dovid";tag=4a482445 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 66]: From: "10000009";tag=as37f9acee [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 10 [ 19]: Content-Length: 184 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 11 [ 0]: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 4 3 IN IP4 212.7.117.61 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 212.7.117.61 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 55500 RTP/AVP 0 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (11 headers 9 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking To) --From tag as37f9acee --To-tag 4a482445 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Acked pending invite 102 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #393 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Stopping retransmission on 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' of Request 102: Match Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP o=- 4 3 IN IP4 212.7.117.61... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '212.7.117.61' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '(null)'. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 212.7.117.61... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found RTP audio format 0 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found RTP audio format 101 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 09:45:45] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x993ce10' [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 212.7.117.61:55500 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0x993cfbc [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0x993cfbc [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Updating call counter for incoming call [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Strict routing enforced for session OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:19010' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '19010'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:19010 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Transmitting (NAT) to 212.7.117.61:19010: ACK sip:dovid@212.7.117.61:19010 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK152c0ae1;rport Max-Forwards: 70 From: "10000009";tag=as37f9acee To: "dovid";tag=4a482445 Contact: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:dov' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Sending pending reinvite on 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Strict routing enforced for session OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:19010' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '19010'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:19010 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Audio is at 5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Initializing already initialized SIP dialog OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (presumably reinvite) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 43]: INVITE sip:dovid@212.7.117.61:19010 SIP/2.0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK533818a6;rport [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 66]: From: "10000009";tag=as37f9acee [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 56]: To: "dovid";tag=4a482445 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 42]: Contact: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Reliably Transmitting (NAT) to 212.7.117.61:19010: INVITE sip:dovid@212.7.117.61:19010 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK533818a6;rport Max-Forwards: 70 From: "10000009";tag=as37f9acee To: "dovid";tag=4a482445 Contact: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.2.2 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: 265 v=0 o=root 1047290220 1047290222 IN IP4 69.167.68.134 s=Asterisk PBX 1.8.2.2 c=IN IP4 69.167.68.134 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #396 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/dovid-00000030' changed end address to 212.7.117.61:55500 (format ulaw) [Jan 26 09:45:45] DEBUG[24077] rtp_engine.c: Oooh, 'SIP/dovid-00000030' was 212.7.117.61:55500/(format unknown) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Sending reinvite on SIP '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' - It's audio soon redirected to IP 212.7.117.61:55500 [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[24077] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[24077] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Audio is at 5060 [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Initializing already initialized SIP dialog 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (presumably reinvite) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 0 [ 46]: INVITE sip:10000009@69.167.68.134:5060 SIP/2.0 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK7c9b9e03 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 2 [ 48]: Route: [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 4 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 5 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 6 [ 39]: Contact: [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 7 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 8 [ 16]: CSeq: 106 INVITE [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 9 [ 32]: User-Agent: Asterisk PBX 1.8.2.2 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] VERBOSE[24077] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060: INVITE sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK7c9b9e03 Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 106 INVITE User-Agent: Asterisk PBX 1.8.2.2 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: 263 v=0 o=root 1043253688 1043253692 IN IP4 212.7.117.61 s=Asterisk PBX 1.8.2.2 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #397 [Jan 26 09:45:45] DEBUG[24077] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/dovid-00000030~ Value: SIP 200 OK Uniqueid: 1296053137.48 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK7c9b9e03 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 106 INVITE Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4033b70000028711d392ffc3f49fc0ba1530b7444c54a8" Server: PBX_MANAGER Content-Length: 0 Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27129 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK7c9b9e03 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 106 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [106]: Proxy-Authenticate: Digest realm="69.167.68.130", nonce="4d4033b70000028711d392ffc3f49fc0ba1530b7444c54a8" [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27129 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (10 headers 0 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Acked pending invite 106 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #397 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Stopping retransmission on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' of Request 106: Match Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 407 to RE-invite on outgoing call 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060: ACK sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK7c9b9e03 Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 106 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Auth attempt 1 on INVITE [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** Our native-bridge filtered capablity: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Audio is at 5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Reliably Transmitting (no NAT) to 69.167.68.130:5060: INVITE sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2590186d Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 107 INVITE User-Agent: Asterisk PBX 1.8.2.2 Proxy-Authorization: Digest username="10000014", realm="69.167.68.130", algorithm=MD5, uri="sip:10000009@69.167.68.134:5060", nonce="4d4033b70000028711d392ffc3f49fc0ba1530b7444c54a8", response="71ffd9791cca8e8d69bd726135528eee" Date: Wed, 26 Jan 2011 14:45:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 1043253688 1043253693 IN IP4 212.7.117.61 s=Asterisk PBX 1.8.2.2 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #398 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 407 Proxy Authentication Required Uniqueid: 1296053137.48 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2590186d From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 107 INVITE Server: PBX_MANAGER Content-Length: 0 Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27127 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 24]: SIP/2.0 100 Giving a try [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2590186d [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 16]: CSeq: 107 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [198]: Warning: 392 69.167.68.130:5060 "Noisy feedback tells: pid=27127 req_src_ip=208.211.92.75 req_src_port=5060 in_uri=sip:10000009@69.167.68.134:5060 out_uri=sip:10000009@69.167.68.134:5060 via_cnt==1" [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (9 headers 0 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: *** SIP TIMER: Cancelling retransmission #398 - INVITE (got response) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Request 107: Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 100 to RE-invite on outgoing call 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 100 Giving a try Uniqueid: 1296053137.48 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2590186d Record-Route: From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 107 INVITE Server: PBX_MANAGER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 276 v=0 o=root 2087990304 2087990306 IN IP4 69.167.68.134 s=PBX_MANAGER c=IN IP4 69.167.68.134 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=direction:active <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2590186d [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 55]: Record-Route: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 54]: From: "dovid" ;tag=as405d8e89 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 47]: To: ;tag=as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 107 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 19]: Server: PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [ 26]: Supported: replaces, timer [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 10 [ 42]: Contact: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 12 [ 19]: Content-Length: 276 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 13 [ 0]: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 1 [ 49]: o=root 2087990304 2087990306 IN IP4 69.167.68.134 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 2 [ 13]: s=PBX_MANAGER [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 3 [ 22]: c=IN IP4 69.167.68.134 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 12022 RTP/AVP 0 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 11 [ 10]: a=sendrecv [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 12 [ 18]: a=direction:active [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (13 headers 13 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking To) --From tag as405d8e89 --To-tag as7f992048 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Acked pending invite 107 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Stopping retransmission on '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' of Request 107: Match Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP o=root 2087990304 2087990306 IN IP4 69.167.68.134... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP s=PBX_MANAGER... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.134' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.134' and port '(null)'. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP c=IN IP4 69.167.68.134... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found RTP audio format 0 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Setting payload 0 based on m type on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found RTP audio format 101 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Setting payload 101 based on m type on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Incorporating payload 0 on 0xb4508100 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Incorporating payload 101 on 0xb4508100 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 26 09:45:45] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd10dec8' [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Peer audio RTP is at port 69.167.68.134:12022 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Copying payload 0 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:45] DEBUG[23737] rtp_engine.c: Copying payload 101 from 0xb4508100 to 0xd10e074 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: We have an owner, now see if we need to change this call [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Updating call counter for outgoing call [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.134:5060' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.134' and port '5060'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 69.167.68.130:5060 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Transmitting (no NAT) to 69.167.68.130:5060: ACK sip:10000009@69.167.68.134:5060 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK334a4f4b Route: Max-Forwards: 70 From: "dovid" ;tag=as405d8e89 To: ;tag=as7f992048 Contact: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 107 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:100' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/fpp-00000031~ Value: SIP 200 OK Uniqueid: 1296053137.48 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK533818a6;rport=5060 Contact: To: "dovid";tag=4a482445 From: "10000009";tag=as37f9acee Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 4 3 IN IP4 212.7.117.61 s=CounterPath eyeBeam 1.5 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK533818a6;rport=5060 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 2 [ 39]: Contact: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 3 [ 56]: To: "dovid";tag=4a482445 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 4 [ 66]: From: "10000009";tag=as37f9acee [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 5 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 103 INVITE [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 9 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 10 [ 19]: Content-Length: 184 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Header 11 [ 0]: [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 4 3 IN IP4 212.7.117.61 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 212.7.117.61 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 55500 RTP/AVP 0 101 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: --- (11 headers 9 lines) --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking To) --From tag as37f9acee --To-tag 4a482445 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Acked pending invite 103 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #396 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Stopping retransmission on 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' of Request 103: Match Found [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Call OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. responded to our reinvite without changing SDP version; ignoring SDP. [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Updating call counter for incoming call [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Strict routing enforced for session OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:45] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:19010' gives... [Jan 26 09:45:45] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '19010'. [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:19010 [Jan 26 09:45:45] VERBOSE[23737] chan_sip.c: Transmitting (NAT) to 212.7.117.61:19010: ACK sip:dovid@212.7.117.61:19010 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK2cfe4625;rport Max-Forwards: 70 From: "10000009";tag=as37f9acee To: "dovid";tag=4a482445 Contact: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:dov' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/dovid-00000030~ Value: SIP 200 OK Uniqueid: 1296053137.48 [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:45] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:45] DEBUG[23750] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 69.167.68.134:12023 OurSSRC: 1043406461 SentNTP: 1296053145.3807485952 SentRTP: 24960 SentPackets: 109 SentOctets: 17440 ReportBlock: FractionLost: 0 CumulativeLoss: 0 IAJitter: 0.0013 TheirLastSR: 0 DLSR: 65507.6570 (sec) [Jan 26 09:45:46] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:46] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:47] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:47] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:48] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:48] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:49] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:49] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:49] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> <-------------> [Jan 26 09:45:49] DEBUG[23737] chan_sip.c: Header 0 [ 0]: [Jan 26 09:45:49] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:49] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:50] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:50] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:50] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:50] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: INVITE [Jan 26 09:45:51] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:69.167.68.130:5060 ---> BYE sip:dovid@208.211.92.75:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 69.167.68.130;branch=z9hG4bKbabb.cc6a2365.0 Via: SIP/2.0/UDP 69.167.68.134:5060;received=69.167.68.134;branch=z9hG4bK7f16073d;rport=5060 Route: Max-Forwards: 69 From: ;tag=as7f992048 To: "dovid" ;tag=as405d8e89 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 102 BYE User-Agent: PBX_MANAGER X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:dovid@208.211.92.75:5060 X-Enswitch-Source: 69.167.68.134:5060 <-------------> [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 0 [ 40]: BYE sip:dovid@208.211.92.75:5060 SIP/2.0 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 1 [ 55]: Record-Route: [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 2 [ 60]: Via: SIP/2.0/UDP 69.167.68.130;branch=z9hG4bKbabb.cc6a2365.0 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 3 [ 92]: Via: SIP/2.0/UDP 69.167.68.134:5060;received=69.167.68.134;branch=z9hG4bK7f16073d;rport=5060 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 4 [ 48]: Route: [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 5 [ 16]: Max-Forwards: 69 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 6 [ 49]: From: ;tag=as7f992048 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 7 [ 52]: To: "dovid" ;tag=as405d8e89 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 8 [ 55]: Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 9 [ 13]: CSeq: 102 BYE [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 10 [ 23]: User-Agent: PBX_MANAGER [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 11 [ 39]: X-Asterisk-HangupCause: Normal Clearing [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 12 [ 30]: X-Asterisk-HangupCauseCode: 16 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 14 [ 45]: X-Enswitch-RURI: sip:dovid@208.211.92.75:5060 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Header 15 [ 37]: X-Enswitch-Source: 69.167.68.134:5060 [Jan 26 09:45:51] VERBOSE[23737] chan_sip.c: --- (16 headers 0 lines) --- [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: = Looking for Call ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 (Checking From) --From tag as7f992048 --To-tag as405d8e89 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Initializing initreq for method BYE - callid 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:51] DEBUG[23737] netsock2.c: Splitting '69.167.68.130' gives... [Jan 26 09:45:51] DEBUG[23737] netsock2.c: ...host '69.167.68.130' and port '(null)'. [Jan 26 09:45:51] VERBOSE[23737] chan_sip.c: Sending to 69.167.68.130:5060 (no NAT) [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Setting SIP_ALREADYGONE on dialog 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:51] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd10dec8' [Jan 26 09:45:51] VERBOSE[23737] chan_sip.c: Scheduling destruction of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' in 32000 ms (Method: BYE) [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Received bye, issuing owner hangup [Jan 26 09:45:51] VERBOSE[23737] chan_sip.c: <--- Transmitting (no NAT) to 69.167.68.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.167.68.130;branch=z9hG4bKbabb.cc6a2365.0;received=69.167.68.130 Via: SIP/2.0/UDP 69.167.68.134:5060;received=69.167.68.134;branch=z9hG4bK7f16073d;rport=5060 Record-Route: From: ;tag=as7f992048 To: "dovid" ;tag=as405d8e89 Call-ID: 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 CSeq: 102 BYE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 69.167.68.130:5060 [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' Method: ACK [Jan 26 09:45:51] DEBUG[23737] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '42a6db4324c2fbb16a275d3417f3e440@69.167.68.130' Method: BYE [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOS Value: ssrc=1043406461;themssrc=1487784032;lp=0;rxjitter=0.001284;rxcount=215;txjitter=0.000000;txcount=109;rlp=0;rtt=0.000000 Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=1043406461;themssrc=1487784032;lp=0;rxjitter=0.001284;rxcount=215;txjitter=0.000000;txcount=109;rlp=0;rtt=0.000000 Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOS Value: ssrc=613198591;themssrc=197475883;lp=5;rxjitter=0.005246;rxcount=358;txjitter=0.000000;txcount=117;rlp=0;rtt=0.174000 Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSBRIDGED Value: ssrc=613198591;themssrc=197475883;lp=5;rxjitter=0.005246;rxcount=358;txjitter=0.000000;txcount=117;rlp=0;rtt=0.174000 Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSJITTERBRIDGED Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSLOSSBRIDGED Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/fpp-00000031 Variable: RTPAUDIOQOSRTTBRIDGED Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1296053138.49 [Jan 26 09:45:51] DEBUG[24077] rtp_engine.c: Oooh, got a hangup [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Sending reinvite on SIP 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' - It's audio soon redirected to IP 208.211.92.75:5060 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Strict routing enforced for session OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:51] VERBOSE[24077] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:51] DEBUG[24077] netsock2.c: Splitting '212.7.117.61:19010' gives... [Jan 26 09:45:51] DEBUG[24077] netsock2.c: ...host '212.7.117.61' and port '19010'. [Jan 26 09:45:51] VERBOSE[24077] chan_sip.c: set_destination: set destination to 212.7.117.61:19010 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 26 09:45:51] VERBOSE[24077] chan_sip.c: Audio is at 5060 [Jan 26 09:45:51] VERBOSE[24077] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 26 09:45:51] VERBOSE[24077] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: -- Done with adding codecs to SDP [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Initializing already initialized SIP dialog OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (presumably reinvite) [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 0 [ 43]: INVITE sip:dovid@212.7.117.61:19010 SIP/2.0 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK49d0cf2c;rport [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 3 [ 66]: From: "10000009";tag=as37f9acee [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 4 [ 56]: To: "dovid";tag=4a482445 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 5 [ 42]: Contact: [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.8.2.2 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 26 09:45:51] VERBOSE[24077] chan_sip.c: Reliably Transmitting (NAT) to 212.7.117.61:19010: INVITE sip:dovid@212.7.117.61:19010 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK49d0cf2c;rport Max-Forwards: 70 From: "10000009";tag=as37f9acee To: "dovid";tag=4a482445 Contact: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.2.2 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: 265 v=0 o=root 1047290220 1047290223 IN IP4 208.211.92.75 s=Asterisk PBX 1.8.2.2 c=IN IP4 208.211.92.75 t=0 0 m=audio 17946 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #400 [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:51] DEBUG[24077] channel.c: Returning from native bridge, channels: SIP/dovid-00000030, SIP/fpp-00000031 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: Unlink Privilege: call,all Channel1: SIP/dovid-00000030 Channel2: SIP/fpp-00000031 Uniqueid1: 1296053137.48 Uniqueid2: 1296053138.49 CallerID1: dovid CallerID2: 10000009 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ANSWEREDTIME Value: 11 Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALEDTIME Value: 14 Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[24077] cdr_mysql.c: Inserting a CDR record. [Jan 26 09:45:51] DEBUG[24077] cdr_mysql.c: SQL command as follows: INSERT INTO asterisk_cdr (calldate,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags) VALUES ('2011-01-26 09:45:37','dovid','10000009','dovid','SIP/dovid-00000030','SIP/fpp-00000031','Dial','SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)F(db_test^1^1)','14','11','ANSWERED','3') [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '2011-01-26 09:45:37' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '"dovid" ' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'dovid' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'SIP/dovid-00000030' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'SIP/fpp-00000031' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'Dial' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)F(db_test^1^1)' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '14' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '11' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'ANSWERED' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is 'DOCUMENTATION' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '(null)' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '1296053137.48' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '(null)' [Jan 26 09:45:51] DEBUG[24077] pbx.c: Function result is '(null)' [Jan 26 09:45:51] DEBUG[24077] cdr_sqlite3_custom.c: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2011-01-26 09:45:37','"dovid" ','dovid','SIP/dovid-00000030','SIP/fpp-00000031','Dial','SIP/10000009@fpp,60,gU(do_dtmf_cc-take-call,s,1)F(db_test^1^1)','14','11','ANSWERED','DOCUMENTATION','','1296053137.48','','') [Jan 26 09:45:51] DEBUG[24077] channel.c: Hanging up channel 'SIP/fpp-00000031' [Jan 26 09:45:51] DEBUG[24077] chan_sip.c: Hangup call SIP/fpp-00000031, SIP callid 42a6db4324c2fbb16a275d3417f3e440@69.167.68.130 [Jan 26 09:45:51] DEBUG[24077] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xd10dec8' [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/fpp-00000031 Uniqueid: 1296053138.49 CallerIDNum: 10000009 CallerIDName: Cause: 16 Cause-txt: Normal Clearing [Jan 26 09:45:51] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - fpp [Jan 26 09:45:51] DEBUG[23716] chan_sip.c: Checking device state for peer fpp [Jan 26 09:45:51] DEBUG[23716] devicestate.c: Changing state for SIP/fpp - state 1 (Not in use) [Jan 26 09:45:51] DEBUG[23716] devicestate.c: device 'SIP/fpp' state '1' [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: DIALSTATUS Value: ANSWER Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: Dial Privilege: call,all SubEvent: End Channel: SIP/dovid-00000030 UniqueID: 1296053137.48 DialStatus: ANSWER [Jan 26 09:45:51] DEBUG[24077] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 26 09:45:51] DEBUG[24077] pbx.c: Launching 'Playback' [Jan 26 09:45:51] VERBOSE[24077] pbx.c: -- Executing [10000009@dovid:3] Playback("SIP/dovid-00000030", "tt-monkeys") in new stack [Jan 26 09:45:51] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/dovid-00000030 Context: dovid Extension: 10000009 Priority: 3 Application: Playback AppData: tt-monkeys Uniqueid: 1296053137.48 [Jan 26 09:45:51] DEBUG[24077] channel.c: Set channel SIP/dovid-00000030 to write format gsm [Jan 26 09:45:51] DEBUG[24077] res_rtp_asterisk.c: Difference is 89768, ms is 11241 [Jan 26 09:45:51] DEBUG[24077] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 26 09:45:51] VERBOSE[24077] file.c: -- Playing 'tt-monkeys.gsm' (language 'en') [Jan 26 09:45:52] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK49d0cf2c;rport=5060 Contact: To: "dovid";tag=4a482445 From: "10000009";tag=as37f9acee Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 184 v=0 o=- 4 3 IN IP4 212.7.117.61 s=CounterPath eyeBeam 1.5 c=IN IP4 212.7.117.61 t=0 0 m=audio 55500 RTP/AVP 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK49d0cf2c;rport=5060 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 2 [ 39]: Contact: [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 3 [ 56]: To: "dovid";tag=4a482445 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 4 [ 66]: From: "10000009";tag=as37f9acee [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 5 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 6 [ 16]: CSeq: 104 INVITE [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 9 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 10 [ 19]: Content-Length: 184 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Header 11 [ 0]: [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 0 [ 3]: v=0 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 1 [ 27]: o=- 4 3 IN IP4 212.7.117.61 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 2 [ 25]: s=CounterPath eyeBeam 1.5 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 3 [ 21]: c=IN IP4 212.7.117.61 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 5 [ 27]: m=audio 55500 RTP/AVP 0 101 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jan 26 09:45:52] VERBOSE[23737] chan_sip.c: --- (11 headers 9 lines) --- [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking To) --From tag as37f9acee --To-tag 4a482445 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Acked pending invite 104 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #400 [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Stopping retransmission on 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' of Request 104: Match Found [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: SIP response 200 to RE-invite on outgoing call OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Call OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. responded to our reinvite without changing SDP version; ignoring SDP. [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Updating call counter for incoming call [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Strict routing enforced for session OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:45:52] VERBOSE[23737] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 26 09:45:52] DEBUG[23737] netsock2.c: Splitting '212.7.117.61:19010' gives... [Jan 26 09:45:52] DEBUG[23737] netsock2.c: ...host '212.7.117.61' and port '19010'. [Jan 26 09:45:52] VERBOSE[23737] chan_sip.c: set_destination: set destination to 212.7.117.61:19010 [Jan 26 09:45:52] VERBOSE[23737] chan_sip.c: Transmitting (NAT) to 212.7.117.61:19010: ACK sip:dovid@212.7.117.61:19010 SIP/2.0 Via: SIP/2.0/UDP 208.211.92.75:5060;branch=z9hG4bK3f9c6aac;rport Max-Forwards: 70 From: "10000009";tag=as37f9acee To: "dovid";tag=4a482445 Contact: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.2.2 Content-Length: 0 --- [Jan 26 09:45:52] DEBUG[23737] chan_sip.c: Trying to put 'ACK sip:dov' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:45:52] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: ~HASH~SIP_CAUSE~SIP/dovid-00000030~ Value: SIP 200 OK Uniqueid: 1296053137.48 [Jan 26 09:45:53] DEBUG[23750] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 212.7.117.61:55501 OurSSRC: 613198591 SentNTP: 1296053153.0250204160 SentRTP: 117928 SentPackets: 176 SentOctets: 28160 ReportBlock: FractionLost: 222 CumulativeLoss: 337 IAJitter: 0.0016 TheirLastSR: 2988237783 DLSR: 8.9390 (sec) [Jan 26 09:45:53] DEBUG[24077] res_rtp_asterisk.c: Got RTCP report of 200 bytes [Jan 26 09:45:53] DEBUG[23750] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 212.7.117.61:55501 PT: 200(Sender Report) ReceptionReports: 2 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 40836 SequenceNumberCycles: 0 IAJitter: 41 LastSR: 45601.0805306368 DLSR: 0.0180(sec) RTT: 171(sec) [Jan 26 09:45:56] DEBUG[24077] res_rtp_asterisk.c: Got RTCP report of 200 bytes [Jan 26 09:45:56] DEBUG[23750] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 212.7.117.61:55501 PT: 200(Sender Report) ReceptionReports: 2 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 40836 SequenceNumberCycles: 0 IAJitter: 41 LastSR: 45601.0805306368 DLSR: 3.0580(sec) RTT: 171(sec) [Jan 26 09:45:58] DEBUG[23750] manager.c: Examining event: Event: RTCPSent Privilege: reporting,all To 212.7.117.61:55501 OurSSRC: 613198591 SentNTP: 1296053158.0250159104 SentRTP: 157928 SentPackets: 426 SentOctets: 68160 ReportBlock: FractionLost: 2 CumulativeLoss: 339 IAJitter: 0.0026 TheirLastSR: 2989035487 DLSR: 1.7700 (sec) [Jan 26 09:45:59] DEBUG[24077] res_rtp_asterisk.c: Got RTCP report of 200 bytes [Jan 26 09:45:59] DEBUG[23750] manager.c: Examining event: Event: RTCPReceived Privilege: reporting,all From 212.7.117.61:55501 PT: 200(Sender Report) ReceptionReports: 2 SenderSSRC: 0 FractionLost: 0 PacketsLost: 0 HighestSequence: 40836 SequenceNumberCycles: 0 IAJitter: 41 LastSR: 45606.0536870912 DLSR: 1.0970(sec) RTT: 174(sec) [Jan 26 09:46:00] DEBUG[24077] res_rtp_asterisk.c: Got RTCP report of 160 bytes [Jan 26 09:46:01] VERBOSE[23737] chan_sip.c: <--- SIP read from UDP:212.7.117.61:19010 ---> BYE sip:10000009@208.211.92.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-182fde1a5b49391e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "10000009";tag=as37f9acee From: "dovid";tag=4a482445 Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 3 BYE User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@208.211.92.75:5060",response="beb8ab810c8e454cb407e4c43eb98288",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 0 [ 43]: BYE sip:10000009@208.211.92.75:5060 SIP/2.0 [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-182fde1a5b49391e-1---d8754z-;rport [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 3 [ 39]: Contact: [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 4 [ 64]: To: "10000009";tag=as37f9acee [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 5 [ 58]: From: "dovid";tag=4a482445 [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 6 [ 53]: Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 7 [ 11]: CSeq: 3 BYE [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 8 [ 45]: User-Agent: eyeBeam release 1102q stamp 51814 [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 9 [168]: Authorization: Digest username="dovid",realm="asterisk",nonce="58bc190d",uri="sip:10000009@208.211.92.75:5060",response="beb8ab810c8e454cb407e4c43eb98288",algorithm=MD5 [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 10 [ 38]: Reason: SIP;description="User Hung Up" [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 26 09:46:01] VERBOSE[23737] chan_sip.c: --- (12 headers 0 lines) --- [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: = Looking for Call ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. (Checking From) --From tag 4a482445 --To-tag as37f9acee [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Initializing initreq for method BYE - callid OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:46:01] DEBUG[23737] netsock2.c: Splitting '192.168.1.10:19010' gives... [Jan 26 09:46:01] DEBUG[23737] netsock2.c: ...host '192.168.1.10' and port '19010'. [Jan 26 09:46:01] VERBOSE[23737] chan_sip.c: Sending to 212.7.117.61:19010 (NAT) [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Setting SIP_ALREADYGONE on dialog OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOS Value: ssrc=613198591;themssrc=197475883;lp=339;rxjitter=0.001193;rxcount=798;txjitter=0.000000;txcount=577;rlp=0;rtt=0.174000 Uniqueid: 1296053137.48 [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:46:01] DEBUG[23737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x993ce10' [Jan 26 09:46:01] VERBOSE[23737] chan_sip.c: Scheduling destruction of SIP dialog 'OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI.' in 32000 ms (Method: BYE) [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Received bye, issuing owner hangup [Jan 26 09:46:01] VERBOSE[23737] chan_sip.c: <--- Transmitting (NAT) to 212.7.117.61:19010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:19010;branch=z9hG4bK-d8754z-182fde1a5b49391e-1---d8754z-;received=212.7.117.61;rport=19010 From: "dovid";tag=4a482445 To: "10000009";tag=as37f9acee Call-ID: OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. CSeq: 3 BYE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 26 09:46:01] DEBUG[23737] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 212.7.117.61:19010 [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSLOSS Value: minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: RTPAUDIOQOSRTT Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000; Uniqueid: 1296053137.48 [Jan 26 09:46:01] DEBUG[24077] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 09:46:01] DEBUG[24077] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 26 09:46:01] DEBUG[24077] channel.c: Set channel SIP/dovid-00000030 to write format ulaw [Jan 26 09:46:01] DEBUG[24077] pbx.c: Spawn extension (dovid,10000009,3) exited non-zero on 'SIP/dovid-00000030' [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/dovid-00000030 Variable: PLAYBACKSTATUS Value: SUCCESS Uniqueid: 1296053137.48 [Jan 26 09:46:01] VERBOSE[24077] pbx.c: == Spawn extension (dovid, 10000009, 3) exited non-zero on 'SIP/dovid-00000030' [Jan 26 09:46:01] DEBUG[24077] channel.c: Soft-Hanging up channel 'SIP/dovid-00000030' [Jan 26 09:46:01] DEBUG[24077] pbx.c: Launching 'NoOp' [Jan 26 09:46:01] VERBOSE[24077] pbx.c: -- Executing [h@dovid:1] NoOp("SIP/dovid-00000030", "ABCDEFGHIJKLMNOPQRSTUVWXYZ") in new stack [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: Newexten Privilege: dialplan,all Channel: SIP/dovid-00000030 Context: dovid Extension: h Priority: 1 Application: NoOp AppData: ABCDEFGHIJKLMNOPQRSTUVWXYZ Uniqueid: 1296053137.48 [Jan 26 09:46:01] DEBUG[24077] channel.c: Hanging up channel 'SIP/dovid-00000030' [Jan 26 09:46:01] DEBUG[24077] chan_sip.c: Hangup call SIP/dovid-00000030, SIP callid OWZkY2Q1M2MyYjExNTliZDFjMzUyNjc5YTBjYWI4ZGI. [Jan 26 09:46:01] DEBUG[24077] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x993ce10' [Jan 26 09:46:01] DEBUG[23750] manager.c: Examining event: Event: Hangup Privilege: call,all Channel: SIP/dovid-00000030 Uniqueid: 1296053137.48 CallerIDNum: dovid CallerIDName: dovid Cause: 16 Cause-txt: Normal Clearing [Jan 26 09:46:01] DEBUG[23716] devicestate.c: No provider found, checking channel drivers for SIP - dovid [Jan 26 09:46:01] DEBUG[23716] chan_sip.c: Checking device state for peer dovid [Jan 26 09:46:01] DEBUG[23716] devicestate.c: Changing state for SIP/dovid - state 1 (Not in use) [Jan 26 09:46:01] DEBUG[23716] devicestate.c: device 'SIP/dovid' state '1'