G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> <-------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4c96c1d7 From: sip:86235@10.66.11.11;tag=d0147b3ef24b612a5158fa3fdb394f84 Call-ID: 4816c0858d83ebb13d0aec054546a0ca@10.66.11.11 CSeq: 2065760989 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK19fa51e2fe235dcdbb06f85cc28cae84 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK19fa51e2fe235dcdbb06f85cc28cae84;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=d0147b3ef24b612a5158fa3fdb394f84 To: sip:8850501@10.144.21.42;tag=as4c96c1d7 Call-ID: 4816c0858d83ebb13d0aec054546a0ca@10.66.11.11 CSeq: 2065760989 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13559' G7-VOIPSERV*CLI> == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101cac30' G7-VOIPSERV*CLI> Really destroying SIP dialog '4816c0858d83ebb13d0aec054546a0ca@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=7497c151254b92ca331c1e0c05d5b0b4 Contact: sip:10.66.11.11 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657829 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK342fcc852168f63b3ea7656f32f7a59e Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244716869 1244716869 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32544 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32544 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32544 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK342fcc852168f63b3ea7656f32f7a59e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7497c151254b92ca331c1e0c05d5b0b4 To: sip:8850501@10.144.21.42 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-1464 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK342fcc852168f63b3ea7656f32f7a59e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7497c151254b92ca331c1e0c05d5b0b4 To: sip:8850501@10.144.21.42;tag=as1d6d4235 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-1464 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13324 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK342fcc852168f63b3ea7656f32f7a59e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7497c151254b92ca331c1e0c05d5b0b4 To: sip:8850501@10.144.21.42;tag=as1d6d4235 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657829 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13324 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1d6d4235 From: "Poste OP2 cmp 2" ;tag=7497c151254b92ca331c1e0c05d5b0b4 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657829 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK507464d562c96fb59ab93d369499ddb1 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1d6d4235 From: sip:86235@10.66.11.11;tag=7497c151254b92ca331c1e0c05d5b0b4 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657830 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK832c20f08ee5d4e62196fd1dec0946f0 Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK832c20f08ee5d4e62196fd1dec0946f0;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=7497c151254b92ca331c1e0c05d5b0b4 To: sip:8850501@10.144.21.42;tag=as1d6d4235 Call-ID: 1b3e99ae21debd20ede5b9e52e870921@10.66.11.11 CSeq: 182657830 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-1464' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '1b3e99ae21debd20ede5b9e52e870921@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=458ba1bdaffcd418c4c4c721c20cb777 Contact: sip:10.66.11.11 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104413 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK103c78c2512bdcc8cd3b32ea794c42f9 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244716873 1244716873 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32560 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32560 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32560 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK103c78c2512bdcc8cd3b32ea794c42f9;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=458ba1bdaffcd418c4c4c721c20cb777 To: sip:8850501@10.144.21.42 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104413 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-15651 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK103c78c2512bdcc8cd3b32ea794c42f9;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=458ba1bdaffcd418c4c4c721c20cb777 To: sip:8850501@10.144.21.42;tag=as29d62fb0 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104413 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-15651 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 12358 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK103c78c2512bdcc8cd3b32ea794c42f9;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=458ba1bdaffcd418c4c4c721c20cb777 To: sip:8850501@10.144.21.42;tag=as29d62fb0 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104413 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12358 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as29d62fb0 From: "Poste OP2 cmp 2" ;tag=458ba1bdaffcd418c4c4c721c20cb777 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104413 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK01fb01a746abb5b09d88cbe3124d1123 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as29d62fb0 From: sip:86235@10.66.11.11;tag=458ba1bdaffcd418c4c4c721c20cb777 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104414 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa79e7f4209ca93081e2c8b358b59474e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa79e7f4209ca93081e2c8b358b59474e;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=458ba1bdaffcd418c4c4c721c20cb777 To: sip:8850501@10.144.21.42;tag=as29d62fb0 Call-ID: 6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11 CSeq: 154104414 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-15651' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '6c5d5a69bb6d984e3bf1493396f83ea5@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=7845c661f17ca70676db840991fc3ba9 Contact: sip:10.66.11.11 Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064975 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8ce73f0b22e918e0cb1e4eb81a5039af Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244716973 1244716973 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32576 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - d134da1632112a6a92f716bc95589b28@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32576 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32576 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8ce73f0b22e918e0cb1e4eb81a5039af;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7845c661f17ca70676db840991fc3ba9 To: sip:8850501@10.144.21.42 Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064975 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-15517 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8ce73f0b22e918e0cb1e4eb81a5039af;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7845c661f17ca70676db840991fc3ba9 To: sip:8850501@10.144.21.42;tag=as2d43419d Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064975 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-15517 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19038 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8ce73f0b22e918e0cb1e4eb81a5039af;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7845c661f17ca70676db840991fc3ba9 To: sip:8850501@10.144.21.42;tag=as2d43419d Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064975 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19038 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as2d43419d From: "Poste OP2 cmp 2" ;tag=7845c661f17ca70676db840991fc3ba9 Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064975 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK41e08a3b91ac23a46cd588d4b7264481 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as2d43419d From: sip:86235@10.66.11.11;tag=7845c661f17ca70676db840991fc3ba9 Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064976 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK88ce183e92a36f345564d068aa516eb7 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK88ce183e92a36f345564d068aa516eb7;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=7845c661f17ca70676db840991fc3ba9 To: sip:8850501@10.144.21.42;tag=as2d43419d Call-ID: d134da1632112a6a92f716bc95589b28@10.66.11.11 CSeq: 1311064976 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-15517' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'd134da1632112a6a92f716bc95589b28@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=78325143c8a3db06f1994bfbfc300117 Contact: sip:10.66.11.11 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461541 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc692e589f0636195c5537e4ae00a4900 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244716978 1244716978 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32592 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32592 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32592 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc692e589f0636195c5537e4ae00a4900;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=78325143c8a3db06f1994bfbfc300117 To: sip:8850501@10.144.21.42 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461541 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101cad68", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5185 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc692e589f0636195c5537e4ae00a4900;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=78325143c8a3db06f1994bfbfc300117 To: sip:8850501@10.144.21.42;tag=as6901e6d4 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461541 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5185 answered SIP/10.66.11.11-101cad68 Audio is at 10.144.21.42 port 10714 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc692e589f0636195c5537e4ae00a4900;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=78325143c8a3db06f1994bfbfc300117 To: sip:8850501@10.144.21.42;tag=as6901e6d4 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461541 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 10714 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6901e6d4 From: "Poste OP2 cmp 2" ;tag=78325143c8a3db06f1994bfbfc300117 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461541 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK64ce17212c2878060f0a65b627c1cdd6 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6901e6d4 From: sip:86235@10.66.11.11;tag=78325143c8a3db06f1994bfbfc300117 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461542 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKec40c75bce658daa598f24c39f13c6af Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKec40c75bce658daa598f24c39f13c6af;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=78325143c8a3db06f1994bfbfc300117 To: sip:8850501@10.144.21.42;tag=as6901e6d4 Call-ID: 8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11 CSeq: 747461542 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5185' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101cad68' Really destroying SIP dialog '8ad8d582a83b83bd6f9f69ae142ec747@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=54dae3c7cccc2fd62561f812a0a98473 Contact: sip:10.66.11.11 Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350268 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK061d6092c3a717e442a8c4ee15469868 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717121 1244717121 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32608 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32608 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) G7-VOIPSERV*CLI> Peer audio RTP is at port 10.144.27.22:32608 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK061d6092c3a717e442a8c4ee15469868;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=54dae3c7cccc2fd62561f812a0a98473 To: sip:8850501@10.144.21.42 Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350268 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-12132 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK061d6092c3a717e442a8c4ee15469868;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=54dae3c7cccc2fd62561f812a0a98473 To: sip:8850501@10.144.21.42;tag=as10616aab Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350268 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-12132 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 19236 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK061d6092c3a717e442a8c4ee15469868;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=54dae3c7cccc2fd62561f812a0a98473 To: sip:8850501@10.144.21.42;tag=as10616aab Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350268 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19236 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as10616aab From: "Poste OP2 cmp 2" ;tag=54dae3c7cccc2fd62561f812a0a98473 Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350268 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa1984ae44b15447c58bb0b9356131054 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as10616aab From: sip:86235@10.66.11.11;tag=54dae3c7cccc2fd62561f812a0a98473 Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350269 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe6336af9dff76cc91df8ed4e7cd08b88 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe6336af9dff76cc91df8ed4e7cd08b88;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=54dae3c7cccc2fd62561f812a0a98473 To: sip:8850501@10.144.21.42;tag=as10616aab Call-ID: 3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11 CSeq: 2105350269 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-12132' G7-VOIPSERV*CLI> == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' G7-VOIPSERV*CLI> Really destroying SIP dialog '3d7c0c4caec4efb7fd95627d51ab9d8a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=ec4857c5c83a4c27e4a779c565ad1462 Contact: sip:10.66.11.11 Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733985 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK394629cdf73df7903fbfdc634126cbb4 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717126 1244717126 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32624 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32624 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32624 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK394629cdf73df7903fbfdc634126cbb4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ec4857c5c83a4c27e4a779c565ad1462 To: sip:8850501@10.144.21.42 Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733985 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13350 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK394629cdf73df7903fbfdc634126cbb4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ec4857c5c83a4c27e4a779c565ad1462 To: sip:8850501@10.144.21.42;tag=as44b26aab Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733985 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13350 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 18648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK394629cdf73df7903fbfdc634126cbb4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ec4857c5c83a4c27e4a779c565ad1462 To: sip:8850501@10.144.21.42;tag=as44b26aab Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733985 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 18648 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as44b26aab From: "Poste OP2 cmp 2" ;tag=ec4857c5c83a4c27e4a779c565ad1462 Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733985 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK38a78e0961272b5e3c7624fa2f9cb3a2 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as44b26aab From: sip:86235@10.66.11.11;tag=ec4857c5c83a4c27e4a779c565ad1462 Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733986 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK60f9f4ce6944f1642da5bc9aecaa27de Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK60f9f4ce6944f1642da5bc9aecaa27de;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=ec4857c5c83a4c27e4a779c565ad1462 To: sip:8850501@10.144.21.42;tag=as44b26aab Call-ID: 27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11 CSeq: 826733986 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Hungup 'IAX2/DSP-IAX-05-13350' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '27b37f2d58d1927ac7952d7a1db3f727@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=a0bec0519d3a348a0359b4e01cb66857 Contact: sip:10.66.11.11 Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266739 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd1fe0accc9c442d9cc4149e71febf47d Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717132 1244717132 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32648 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32648 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) G7-VOIPSERV*CLI> Peer audio RTP is at port 10.144.27.22:32648 G7-VOIPSERV*CLI> Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd1fe0accc9c442d9cc4149e71febf47d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=a0bec0519d3a348a0359b4e01cb66857 To: sip:8850501@10.144.21.42 Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266739 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-9550 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd1fe0accc9c442d9cc4149e71febf47d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=a0bec0519d3a348a0359b4e01cb66857 To: sip:8850501@10.144.21.42;tag=as01fc527a Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266739 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-9550 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 19384 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd1fe0accc9c442d9cc4149e71febf47d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=a0bec0519d3a348a0359b4e01cb66857 To: sip:8850501@10.144.21.42;tag=as01fc527a Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266739 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19384 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as01fc527a From: "Poste OP2 cmp 2" ;tag=a0bec0519d3a348a0359b4e01cb66857 Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266739 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK18c62f9150c111e16b9aaa71a2d6e794 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as01fc527a From: sip:86235@10.66.11.11;tag=a0bec0519d3a348a0359b4e01cb66857 Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266740 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK76321ea6d2afd91ea83c10e6edc4d049 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK76321ea6d2afd91ea83c10e6edc4d049;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=a0bec0519d3a348a0359b4e01cb66857 To: sip:8850501@10.144.21.42;tag=as01fc527a Call-ID: f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11 CSeq: 351266740 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-9550' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'f4a0fd0b6ebd3a77453cdc903967d847@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=c738f2b6ff613d36166c1d6269e0f74b Contact: sip:10.66.11.11 Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279587 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb3ea562f56f630b3dbd94c5e2e3dc1c5 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717293 1244717293 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32664 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 G7-VOIPSERV*CLI> Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32664 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32664 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb3ea562f56f630b3dbd94c5e2e3dc1c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c738f2b6ff613d36166c1d6269e0f74b To: sip:8850501@10.144.21.42 Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279587 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-6087 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb3ea562f56f630b3dbd94c5e2e3dc1c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c738f2b6ff613d36166c1d6269e0f74b To: sip:8850501@10.144.21.42;tag=as4eb5d0dd Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279587 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-6087 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 17096 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb3ea562f56f630b3dbd94c5e2e3dc1c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c738f2b6ff613d36166c1d6269e0f74b To: sip:8850501@10.144.21.42;tag=as4eb5d0dd Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279587 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17096 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4eb5d0dd From: "Poste OP2 cmp 2" ;tag=c738f2b6ff613d36166c1d6269e0f74b Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279587 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK375793b3f679c35690e102e54b1af316 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4eb5d0dd From: sip:86235@10.66.11.11;tag=c738f2b6ff613d36166c1d6269e0f74b Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279588 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7e11762764bc67109ce744b087896c54 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7e11762764bc67109ce744b087896c54;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=c738f2b6ff613d36166c1d6269e0f74b To: sip:8850501@10.144.21.42;tag=as4eb5d0dd Call-ID: 5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11 CSeq: 858279588 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-6087' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '5960249a0c9f1f0db4ecf8967eb53aa6@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=4f02793049e86062b39bafe6e9732185 Contact: sip:10.66.11.11 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800382 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9cef3e483cca4aabf0781bae34a808bd Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717296 1244717296 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32680 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32680 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32680 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9cef3e483cca4aabf0781bae34a808bd;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=4f02793049e86062b39bafe6e9732185 To: sip:8850501@10.144.21.42 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800382 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14605 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9cef3e483cca4aabf0781bae34a808bd;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=4f02793049e86062b39bafe6e9732185 To: sip:8850501@10.144.21.42;tag=as0b2d0d39 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800382 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14605 answered SIP/10.66.11.11-101d5c48 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 16096 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9cef3e483cca4aabf0781bae34a808bd;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=4f02793049e86062b39bafe6e9732185 To: sip:8850501@10.144.21.42;tag=as0b2d0d39 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800382 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16096 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0b2d0d39 From: "Poste OP2 cmp 2" ;tag=4f02793049e86062b39bafe6e9732185 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800382 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd6376dd06549b7044addc38c212e5be4 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0b2d0d39 From: sip:86235@10.66.11.11;tag=4f02793049e86062b39bafe6e9732185 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800383 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2fe5a45566465d25509a183c0e5cf712 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2fe5a45566465d25509a183c0e5cf712;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=4f02793049e86062b39bafe6e9732185 To: sip:8850501@10.144.21.42;tag=as0b2d0d39 Call-ID: 40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11 CSeq: 1562800383 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14605' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '40a6e6c447e0103b2f838ae3d1e6eda6@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=e03a12faf18cb3705998f3c02a51c6fe Contact: sip:10.66.11.11 Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076009 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdf3e785b172d13156611322ba5674e90 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717300 1244717300 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32704 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b0dad32493f0b976a989023cbf317bb8@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32704 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32704 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdf3e785b172d13156611322ba5674e90;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=e03a12faf18cb3705998f3c02a51c6fe To: sip:8850501@10.144.21.42 Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076009 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13794 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdf3e785b172d13156611322ba5674e90;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=e03a12faf18cb3705998f3c02a51c6fe To: sip:8850501@10.144.21.42;tag=as5acadf00 Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076009 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13794 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 10446 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdf3e785b172d13156611322ba5674e90;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=e03a12faf18cb3705998f3c02a51c6fe To: sip:8850501@10.144.21.42;tag=as5acadf00 Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076009 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 10446 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as5acadf00 From: "Poste OP2 cmp 2" ;tag=e03a12faf18cb3705998f3c02a51c6fe Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076009 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7b5471fc24bf2ee67ea8683a67c3a5f4 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as5acadf00 From: sip:86235@10.66.11.11;tag=e03a12faf18cb3705998f3c02a51c6fe Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076010 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6ab9df84f1e4dcb4f737e41277c5be81 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6ab9df84f1e4dcb4f737e41277c5be81;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=e03a12faf18cb3705998f3c02a51c6fe To: sip:8850501@10.144.21.42;tag=as5acadf00 Call-ID: b0dad32493f0b976a989023cbf317bb8@10.66.11.11 CSeq: 807076010 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13794' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'b0dad32493f0b976a989023cbf317bb8@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=0e9162cd8aadd327a45b5f72a79a82b0 Contact: sip:10.66.11.11 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368981 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81cdb15ccdc95f3aec53987e70ebdc8 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717337 1244717337 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32728 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32728 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32728 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81cdb15ccdc95f3aec53987e70ebdc8;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0e9162cd8aadd327a45b5f72a79a82b0 To: sip:8850501@10.144.21.42 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368981 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-1235 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81cdb15ccdc95f3aec53987e70ebdc8;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0e9162cd8aadd327a45b5f72a79a82b0 To: sip:8850501@10.144.21.42;tag=as710b0948 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368981 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-1235 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14548 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81cdb15ccdc95f3aec53987e70ebdc8;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0e9162cd8aadd327a45b5f72a79a82b0 To: sip:8850501@10.144.21.42;tag=as710b0948 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368981 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14548 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as710b0948 From: "Poste OP2 cmp 2" ;tag=0e9162cd8aadd327a45b5f72a79a82b0 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368981 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbdd8d292f488d0b2c61f285a7c83a8cd Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as710b0948 From: sip:86235@10.66.11.11;tag=0e9162cd8aadd327a45b5f72a79a82b0 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368982 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3c38d8425ba31a2b5e8db1bacca9e985 Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3c38d8425ba31a2b5e8db1bacca9e985;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=0e9162cd8aadd327a45b5f72a79a82b0 To: sip:8850501@10.144.21.42;tag=as710b0948 Call-ID: a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11 CSeq: 818368982 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-1235' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'a8ec4d8ced0d1890547b79074a8a5218@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=c00a20ed4aeb58f5eed5fec0929da846 Contact: sip:10.66.11.11 Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822322 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK74bcee7121dbe058d80dadaf19d6a056 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717341 1244717341 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32544 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - def2c384034f6c39a64899691e9d77d6@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32544 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32544 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK74bcee7121dbe058d80dadaf19d6a056;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c00a20ed4aeb58f5eed5fec0929da846 To: sip:8850501@10.144.21.42 Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822322 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-603 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK74bcee7121dbe058d80dadaf19d6a056;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c00a20ed4aeb58f5eed5fec0929da846 To: sip:8850501@10.144.21.42;tag=as1c32a2bc Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822322 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-603 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14656 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK74bcee7121dbe058d80dadaf19d6a056;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c00a20ed4aeb58f5eed5fec0929da846 To: sip:8850501@10.144.21.42;tag=as1c32a2bc Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822322 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14656 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1c32a2bc From: "Poste OP2 cmp 2" ;tag=c00a20ed4aeb58f5eed5fec0929da846 Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822322 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc68ab42708e6c1d3928a1921cfd3afec Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1c32a2bc From: sip:86235@10.66.11.11;tag=c00a20ed4aeb58f5eed5fec0929da846 Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822323 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK343ed887730672de89115e1e8eda2c88 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK343ed887730672de89115e1e8eda2c88;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=c00a20ed4aeb58f5eed5fec0929da846 To: sip:8850501@10.144.21.42;tag=as1c32a2bc Call-ID: def2c384034f6c39a64899691e9d77d6@10.66.11.11 CSeq: 1545822323 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-603' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'def2c384034f6c39a64899691e9d77d6@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=85ba268d3bd92e0493973d8f1dc1fba1 Contact: sip:10.66.11.11 Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906602 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb5947467695c23234f39da5677c5dd76 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717347 1244717347 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32560 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32560 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32560 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb5947467695c23234f39da5677c5dd76;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=85ba268d3bd92e0493973d8f1dc1fba1 To: sip:8850501@10.144.21.42 Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906602 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13483 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb5947467695c23234f39da5677c5dd76;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=85ba268d3bd92e0493973d8f1dc1fba1 To: sip:8850501@10.144.21.42;tag=as2a996dbf Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906602 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13483 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 15416 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb5947467695c23234f39da5677c5dd76;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=85ba268d3bd92e0493973d8f1dc1fba1 To: sip:8850501@10.144.21.42;tag=as2a996dbf Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906602 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 15416 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as2a996dbf From: "Poste OP2 cmp 2" ;tag=85ba268d3bd92e0493973d8f1dc1fba1 Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906602 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK35e5c87783e7576a76363fa20870004a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as2a996dbf From: sip:86235@10.66.11.11;tag=85ba268d3bd92e0493973d8f1dc1fba1 Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906603 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe3f90dd776b41f445c46afce2761046f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe3f90dd776b41f445c46afce2761046f;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=85ba268d3bd92e0493973d8f1dc1fba1 To: sip:8850501@10.144.21.42;tag=as2a996dbf Call-ID: a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11 CSeq: 186906603 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13483' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'a58c4aecbddea0c85d60e29684e4ad8b@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=956434bd47fac86a05645797c292089b Contact: sip:10.66.11.11 Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572535 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKddb917a3ca5e8c784b802a754be856a0 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717351 1244717351 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32576 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 8a505b0eba700513f64e79d88abdf768@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32576 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32576 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKddb917a3ca5e8c784b802a754be856a0;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=956434bd47fac86a05645797c292089b To: sip:8850501@10.144.21.42 Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572535 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-12720 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKddb917a3ca5e8c784b802a754be856a0;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=956434bd47fac86a05645797c292089b To: sip:8850501@10.144.21.42;tag=as714e9123 Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572535 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-12720 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13970 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKddb917a3ca5e8c784b802a754be856a0;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=956434bd47fac86a05645797c292089b To: sip:8850501@10.144.21.42;tag=as714e9123 Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572535 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13970 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as714e9123 From: "Poste OP2 cmp 2" ;tag=956434bd47fac86a05645797c292089b Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572535 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7455e9f9033f69e063032306155da13d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as714e9123 From: sip:86235@10.66.11.11;tag=956434bd47fac86a05645797c292089b Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572536 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK46f12429c6d91a2e966e78bce6572d8a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK46f12429c6d91a2e966e78bce6572d8a;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=956434bd47fac86a05645797c292089b To: sip:8850501@10.144.21.42;tag=as714e9123 Call-ID: 8a505b0eba700513f64e79d88abdf768@10.66.11.11 CSeq: 754572536 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-12720' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '8a505b0eba700513f64e79d88abdf768@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=2e16b55497a41298887a265642ce1ef7 Contact: sip:10.66.11.11 Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270480 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7ac4f6fa8ab3a743fb66fcb280919e6a Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717354 1244717354 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32592 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> G7-VOIPSERV*CLI> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - fbd493b9661832dda31811fed3f33425@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32592 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32592 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7ac4f6fa8ab3a743fb66fcb280919e6a;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2e16b55497a41298887a265642ce1ef7 To: sip:8850501@10.144.21.42 Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270480 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-7504 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7ac4f6fa8ab3a743fb66fcb280919e6a;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2e16b55497a41298887a265642ce1ef7 To: sip:8850501@10.144.21.42;tag=as6799b41b Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270480 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-7504 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 17070 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7ac4f6fa8ab3a743fb66fcb280919e6a;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2e16b55497a41298887a265642ce1ef7 To: sip:8850501@10.144.21.42;tag=as6799b41b Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270480 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17070 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6799b41b From: "Poste OP2 cmp 2" ;tag=2e16b55497a41298887a265642ce1ef7 Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270480 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf5f22bcb66cc2eaee4c2e4b5f673a645 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6799b41b From: sip:86235@10.66.11.11;tag=2e16b55497a41298887a265642ce1ef7 Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270481 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc187eabb3da6ead45eb483d6315032fb Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc187eabb3da6ead45eb483d6315032fb;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=2e16b55497a41298887a265642ce1ef7 To: sip:8850501@10.144.21.42;tag=as6799b41b Call-ID: fbd493b9661832dda31811fed3f33425@10.66.11.11 CSeq: 916270481 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-7504' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'fbd493b9661832dda31811fed3f33425@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 Contact: sip:10.66.11.11 Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6652d32d1d07bdf34f66799d3ec13aeb Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717360 1244717360 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32608 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32608 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32608 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6652d32d1d07bdf34f66799d3ec13aeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 To: sip:8850501@10.144.21.42 Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5506 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6652d32d1d07bdf34f66799d3ec13aeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 To: sip:8850501@10.144.21.42;tag=as0f11f12f Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5506 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13948 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6652d32d1d07bdf34f66799d3ec13aeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 To: sip:8850501@10.144.21.42;tag=as0f11f12f Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13948 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0f11f12f From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1e114e0c0ae85ad01a3f4b0a92482ccb Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Nov 30 01:00:09] ERROR[464]: chan_sip.c:15553 sipsock_read: We could NOT get the channel lock for SIP/10.66.11.11-101d5c48! [Nov 30 01:00:09] ERROR[464]: chan_sip.c:15554 sipsock_read: SIP transaction failed: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 G7-VOIPSERV*CLI> Retransmitting #1 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6652d32d1d07bdf34f66799d3ec13aeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 To: sip:8850501@10.144.21.42;tag=as0f11f12f Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13948 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0f11f12f From: "Poste OP2 cmp 2" ;tag=9a7c360f31c27b2f6a76cdda2294c087 Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436207 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1e114e0c0ae85ad01a3f4b0a92482ccb Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0f11f12f From: sip:86235@10.66.11.11;tag=9a7c360f31c27b2f6a76cdda2294c087 Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436208 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0b727e262750aa22d0a8e823196a067d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0b727e262750aa22d0a8e823196a067d;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=9a7c360f31c27b2f6a76cdda2294c087 To: sip:8850501@10.144.21.42;tag=as0f11f12f Call-ID: b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11 CSeq: 1920436208 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5506' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'b5198728a47a0a7e0dbf72613b1b5254@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=0a18de08275eb6917f8c688ad8f8682e Contact: sip:10.66.11.11 Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391553 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2ea1bc08dadb631837d623ee0c1f2810 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717365 1244717365 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32624 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 06708af734a2f3964986900687cc480a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32624 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32624 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2ea1bc08dadb631837d623ee0c1f2810;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0a18de08275eb6917f8c688ad8f8682e To: sip:8850501@10.144.21.42 Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391553 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5243 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2ea1bc08dadb631837d623ee0c1f2810;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0a18de08275eb6917f8c688ad8f8682e To: sip:8850501@10.144.21.42;tag=as340a1b9e Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391553 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5243 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13090 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2ea1bc08dadb631837d623ee0c1f2810;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0a18de08275eb6917f8c688ad8f8682e To: sip:8850501@10.144.21.42;tag=as340a1b9e Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391553 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13090 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as340a1b9e From: "Poste OP2 cmp 2" ;tag=0a18de08275eb6917f8c688ad8f8682e Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391553 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8dec3082c0ad23dfce1cadcf19b1c51 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as340a1b9e From: sip:86235@10.66.11.11;tag=0a18de08275eb6917f8c688ad8f8682e Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391554 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf2ff06fbe1467bd582da37290522ee91 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf2ff06fbe1467bd582da37290522ee91;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=0a18de08275eb6917f8c688ad8f8682e To: sip:8850501@10.144.21.42;tag=as340a1b9e Call-ID: 06708af734a2f3964986900687cc480a@10.66.11.11 CSeq: 1324391554 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Hungup 'IAX2/DSP-IAX-05-5243' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '06708af734a2f3964986900687cc480a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=18f7acff8165b8ba1020c7902309f79b Contact: sip:10.66.11.11 Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957491 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd26ae6eb5d8696f9249432cea17d6212 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717453 1244717453 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32648 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32648 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32648 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd26ae6eb5d8696f9249432cea17d6212;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=18f7acff8165b8ba1020c7902309f79b To: sip:8850501@10.144.21.42 Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957491 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-1913 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd26ae6eb5d8696f9249432cea17d6212;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=18f7acff8165b8ba1020c7902309f79b To: sip:8850501@10.144.21.42;tag=as137c87dd Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957491 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-1913 answered SIP/10.66.11.11-101d0ac8 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 19140 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd26ae6eb5d8696f9249432cea17d6212;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=18f7acff8165b8ba1020c7902309f79b To: sip:8850501@10.144.21.42;tag=as137c87dd Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957491 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19140 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as137c87dd From: "Poste OP2 cmp 2" ;tag=18f7acff8165b8ba1020c7902309f79b Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957491 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK934c3577d42e233387901eb527c11fd7 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as137c87dd From: sip:86235@10.66.11.11;tag=18f7acff8165b8ba1020c7902309f79b Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957492 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKec2b61145dca391a3fcb0f419fe9f403 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKec2b61145dca391a3fcb0f419fe9f403;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=18f7acff8165b8ba1020c7902309f79b To: sip:8850501@10.144.21.42;tag=as137c87dd Call-ID: b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11 CSeq: 664957492 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-1913' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'b438f11834be651dd9cb6c6d49b0ccf0@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=586992ce00286c647e13fefc4257b247 Contact: sip:10.66.11.11 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442450 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4e0ed20ea341a0c6790ea5f998178cef Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717457 1244717457 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32664 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32664 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32664 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4e0ed20ea341a0c6790ea5f998178cef;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=586992ce00286c647e13fefc4257b247 To: sip:8850501@10.144.21.42 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442450 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-3127 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4e0ed20ea341a0c6790ea5f998178cef;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=586992ce00286c647e13fefc4257b247 To: sip:8850501@10.144.21.42;tag=as26af11c5 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442450 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-3127 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 15448 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4e0ed20ea341a0c6790ea5f998178cef;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=586992ce00286c647e13fefc4257b247 To: sip:8850501@10.144.21.42;tag=as26af11c5 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442450 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 15448 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as26af11c5 From: "Poste OP2 cmp 2" ;tag=586992ce00286c647e13fefc4257b247 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442450 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK842635233d6332351ea72efffcae3870 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as26af11c5 From: sip:86235@10.66.11.11;tag=586992ce00286c647e13fefc4257b247 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442451 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK95fe1d922954a4760eb4e35d877e462f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK95fe1d922954a4760eb4e35d877e462f;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=586992ce00286c647e13fefc4257b247 To: sip:8850501@10.144.21.42;tag=as26af11c5 Call-ID: a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11 CSeq: 1609442451 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-3127' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'a83b5b96d479be8b6f046fa7a9936dd6@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=0021b8427f72baae4c66f0754e809f19 Contact: sip:10.66.11.11 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414791 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe7a6614a2097086fce2c12bad2a5bd16 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717461 1244717461 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32680 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32680 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32680 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe7a6614a2097086fce2c12bad2a5bd16;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0021b8427f72baae4c66f0754e809f19 To: sip:8850501@10.144.21.42 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414791 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-22 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe7a6614a2097086fce2c12bad2a5bd16;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0021b8427f72baae4c66f0754e809f19 To: sip:8850501@10.144.21.42;tag=as4f24eb60 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414791 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-22 answered SIP/10.66.11.11-101d0ac8 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 12858 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe7a6614a2097086fce2c12bad2a5bd16;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0021b8427f72baae4c66f0754e809f19 To: sip:8850501@10.144.21.42;tag=as4f24eb60 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414791 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12858 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4f24eb60 From: "Poste OP2 cmp 2" ;tag=0021b8427f72baae4c66f0754e809f19 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414791 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa07e599b999c37c2a66d731e5c58f32a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4f24eb60 From: sip:86235@10.66.11.11;tag=0021b8427f72baae4c66f0754e809f19 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414792 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20a05bed2457565d848d2d4c2e224b16 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20a05bed2457565d848d2d4c2e224b16;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=0021b8427f72baae4c66f0754e809f19 To: sip:8850501@10.144.21.42;tag=as4f24eb60 Call-ID: 599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11 CSeq: 176414792 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-22' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '599d0e5fa6c4f53bb01105995c20aca5@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=de0b9c6fcf3f30b87882b14f4ee121a8 Contact: sip:10.66.11.11 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932427 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe6fc0f976d3223bcb0879cd1058a688f Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717467 1244717467 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32704 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32704 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32704 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe6fc0f976d3223bcb0879cd1058a688f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=de0b9c6fcf3f30b87882b14f4ee121a8 To: sip:8850501@10.144.21.42 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932427 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-11010 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe6fc0f976d3223bcb0879cd1058a688f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=de0b9c6fcf3f30b87882b14f4ee121a8 To: sip:8850501@10.144.21.42;tag=as073cbc05 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932427 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-11010 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 16344 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe6fc0f976d3223bcb0879cd1058a688f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=de0b9c6fcf3f30b87882b14f4ee121a8 To: sip:8850501@10.144.21.42;tag=as073cbc05 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932427 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16344 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as073cbc05 From: "Poste OP2 cmp 2" ;tag=de0b9c6fcf3f30b87882b14f4ee121a8 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932427 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3fcf97f6a24e381f351ef09a79212201 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as073cbc05 From: sip:86235@10.66.11.11;tag=de0b9c6fcf3f30b87882b14f4ee121a8 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932428 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7f739ea192a5e2382155ae0c05a996a7 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7f739ea192a5e2382155ae0c05a996a7;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=de0b9c6fcf3f30b87882b14f4ee121a8 To: sip:8850501@10.144.21.42;tag=as073cbc05 Call-ID: 9a6a5f39bf479eeee9473ec824c991df@10.66.11.11 CSeq: 1158932428 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-11010' G7-VOIPSERV*CLI> == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' G7-VOIPSERV*CLI> Really destroying SIP dialog '9a6a5f39bf479eeee9473ec824c991df@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=81cb311365f253e68e3199f3d17d9a52 Contact: sip:10.66.11.11 Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792743 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6eb0bcbbcc826b73de9e9d2508bc7b86 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717470 1244717470 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32728 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b2c781910c945df012e79bd01243dd4a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32728 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32728 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6eb0bcbbcc826b73de9e9d2508bc7b86;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=81cb311365f253e68e3199f3d17d9a52 To: sip:8850501@10.144.21.42 Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792743 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-608 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6eb0bcbbcc826b73de9e9d2508bc7b86;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=81cb311365f253e68e3199f3d17d9a52 To: sip:8850501@10.144.21.42;tag=as44c079cc Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792743 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-608 answered SIP/10.66.11.11-101d0ac8 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 10652 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6eb0bcbbcc826b73de9e9d2508bc7b86;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=81cb311365f253e68e3199f3d17d9a52 To: sip:8850501@10.144.21.42;tag=as44c079cc Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792743 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 10652 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as44c079cc From: "Poste OP2 cmp 2" ;tag=81cb311365f253e68e3199f3d17d9a52 Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792743 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf4d0c691d189a59f1ed8fd6802b2cbe1 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as44c079cc From: sip:86235@10.66.11.11;tag=81cb311365f253e68e3199f3d17d9a52 Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792744 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKab9bf7c02ccdce4f9b33e10e980c5430 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKab9bf7c02ccdce4f9b33e10e980c5430;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=81cb311365f253e68e3199f3d17d9a52 To: sip:8850501@10.144.21.42;tag=as44c079cc Call-ID: b2c781910c945df012e79bd01243dd4a@10.66.11.11 CSeq: 71792744 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-608' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'b2c781910c945df012e79bd01243dd4a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=cf9535f6105556df029a979c9afe11af Contact: sip:10.66.11.11 Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470276 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK49c6513939036ec4f550398b26fa1215 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717473 1244717473 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32544 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32544 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32544 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK49c6513939036ec4f550398b26fa1215;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=cf9535f6105556df029a979c9afe11af To: sip:8850501@10.144.21.42 Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470276 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-9283 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK49c6513939036ec4f550398b26fa1215;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=cf9535f6105556df029a979c9afe11af To: sip:8850501@10.144.21.42;tag=as7968b99d Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470276 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-9283 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 12730 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK49c6513939036ec4f550398b26fa1215;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=cf9535f6105556df029a979c9afe11af To: sip:8850501@10.144.21.42;tag=as7968b99d Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470276 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12730 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7968b99d From: "Poste OP2 cmp 2" ;tag=cf9535f6105556df029a979c9afe11af Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470276 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK33eab3b4c42553bf69fc550047ecb152 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7968b99d From: sip:86235@10.66.11.11;tag=cf9535f6105556df029a979c9afe11af Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470277 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd2d032661fdd12aed1d9aade570eda65 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd2d032661fdd12aed1d9aade570eda65;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=cf9535f6105556df029a979c9afe11af To: sip:8850501@10.144.21.42;tag=as7968b99d Call-ID: d8bf538f73186fb08a0b837d5bd57c54@10.66.11.11 CSeq: 2086470277 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-9283' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'd8bf538f73186fb08a0b837d5bd57c54@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=709a2b2f948635a9924c0b3298f0b156 Contact: sip:10.66.11.11 Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899053 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0555d3deb271d0b1d55a74339a456ded Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717475 1244717475 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32560 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32560 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32560 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0555d3deb271d0b1d55a74339a456ded;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=709a2b2f948635a9924c0b3298f0b156 To: sip:8850501@10.144.21.42 Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899053 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14053 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0555d3deb271d0b1d55a74339a456ded;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=709a2b2f948635a9924c0b3298f0b156 To: sip:8850501@10.144.21.42;tag=as4f467ffb Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899053 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14053 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19830 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0555d3deb271d0b1d55a74339a456ded;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=709a2b2f948635a9924c0b3298f0b156 To: sip:8850501@10.144.21.42;tag=as4f467ffb Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899053 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19830 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4f467ffb From: "Poste OP2 cmp 2" ;tag=709a2b2f948635a9924c0b3298f0b156 Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899053 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK992eb6928b382e8c667fe211a20573ec Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4f467ffb From: sip:86235@10.66.11.11;tag=709a2b2f948635a9924c0b3298f0b156 Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899054 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa037081af7931b6b4b25c4b19095aebd Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa037081af7931b6b4b25c4b19095aebd;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=709a2b2f948635a9924c0b3298f0b156 To: sip:8850501@10.144.21.42;tag=as4f467ffb Call-ID: a201a02b75d54bd794fef1b86dfe370c@10.66.11.11 CSeq: 1064899054 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14053' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'a201a02b75d54bd794fef1b86dfe370c@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f Contact: sip:10.66.11.11 Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144941 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7a07b2346936cc12fb056f972997dd6f Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717478 1244717478 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32576 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32576 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32576 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7a07b2346936cc12fb056f972997dd6f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f To: sip:8850501@10.144.21.42 Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144941 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-7505 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7a07b2346936cc12fb056f972997dd6f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f To: sip:8850501@10.144.21.42;tag=as277b8750 Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144941 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-7505 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13048 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7a07b2346936cc12fb056f972997dd6f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f To: sip:8850501@10.144.21.42;tag=as277b8750 Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144941 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13048 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as277b8750 From: "Poste OP2 cmp 2" ;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144941 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf44f48c9646770f62cdfb3cbc43943eb Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as277b8750 From: sip:86235@10.66.11.11;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144942 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5d94c619fffacfe56c30eff313970f9a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5d94c619fffacfe56c30eff313970f9a;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=d267ab35e4ad8a2ee3f4c9bc951e3a4f To: sip:8850501@10.144.21.42;tag=as277b8750 Call-ID: 26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11 CSeq: 1809144942 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-7505' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '26e84b6886f3c5f5bf294c92ba97c1da@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=70e3e34808edb5ddc3ac98b23dec782d Contact: sip:10.66.11.11 Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784381 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20e0ca90c428b9b3fd766adf72d9e077 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717481 1244717481 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32592 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> G7-VOIPSERV*CLI> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32592 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32592 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20e0ca90c428b9b3fd766adf72d9e077;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=70e3e34808edb5ddc3ac98b23dec782d To: sip:8850501@10.144.21.42 Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784381 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-15528 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20e0ca90c428b9b3fd766adf72d9e077;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=70e3e34808edb5ddc3ac98b23dec782d To: sip:8850501@10.144.21.42;tag=as7748c72d Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784381 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-15528 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14998 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20e0ca90c428b9b3fd766adf72d9e077;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=70e3e34808edb5ddc3ac98b23dec782d To: sip:8850501@10.144.21.42;tag=as7748c72d Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784381 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14998 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7748c72d From: "Poste OP2 cmp 2" ;tag=70e3e34808edb5ddc3ac98b23dec782d Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784381 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9c498caedabf84536ce1f667c029b8b8 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7748c72d From: sip:86235@10.66.11.11;tag=70e3e34808edb5ddc3ac98b23dec782d Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784382 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK447cead9133c659ea45f92f56a6782da Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK447cead9133c659ea45f92f56a6782da;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=70e3e34808edb5ddc3ac98b23dec782d To: sip:8850501@10.144.21.42;tag=as7748c72d Call-ID: 51dbe08ee17d22e991c603e517c5b483@10.66.11.11 CSeq: 1588784382 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-15528' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '51dbe08ee17d22e991c603e517c5b483@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=65882c6cc4349b712b829cb707ec1d40 Contact: sip:10.66.11.11 Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526177 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfc407255a85b197d7d48c33d184045fb Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717484 1244717484 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32608 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32608 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32608 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfc407255a85b197d7d48c33d184045fb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=65882c6cc4349b712b829cb707ec1d40 To: sip:8850501@10.144.21.42 Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526177 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5340 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfc407255a85b197d7d48c33d184045fb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=65882c6cc4349b712b829cb707ec1d40 To: sip:8850501@10.144.21.42;tag=as126d43fa Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526177 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5340 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 10392 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfc407255a85b197d7d48c33d184045fb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=65882c6cc4349b712b829cb707ec1d40 To: sip:8850501@10.144.21.42;tag=as126d43fa Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526177 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 10392 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as126d43fa From: "Poste OP2 cmp 2" ;tag=65882c6cc4349b712b829cb707ec1d40 Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526177 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK77a7cfa71f45b5a1109e7b9a6a0aae5d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as126d43fa From: sip:86235@10.66.11.11;tag=65882c6cc4349b712b829cb707ec1d40 Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526178 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0d8c8b9c52577c407808f4f9af522bb2 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0d8c8b9c52577c407808f4f9af522bb2;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=65882c6cc4349b712b829cb707ec1d40 To: sip:8850501@10.144.21.42;tag=as126d43fa Call-ID: 605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11 CSeq: 1776526178 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5340' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '605111856a8c3a2faa3f7ed7e1741aff@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=eae726c9592cbf633ada592803260761 Contact: sip:10.66.11.11 Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809507 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8f7738f42f325a3ef8a299324a99372b Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717486 1244717486 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32624 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b034b607fe136cc347a3edea69432342@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32624 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32624 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8f7738f42f325a3ef8a299324a99372b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=eae726c9592cbf633ada592803260761 To: sip:8850501@10.144.21.42 Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809507 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5044 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8f7738f42f325a3ef8a299324a99372b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=eae726c9592cbf633ada592803260761 To: sip:8850501@10.144.21.42;tag=as6739302b Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809507 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5044 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19144 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8f7738f42f325a3ef8a299324a99372b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=eae726c9592cbf633ada592803260761 To: sip:8850501@10.144.21.42;tag=as6739302b Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809507 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19144 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6739302b From: "Poste OP2 cmp 2" ;tag=eae726c9592cbf633ada592803260761 Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809507 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa73a7270482c3d34785716436ac78db5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6739302b From: sip:86235@10.66.11.11;tag=eae726c9592cbf633ada592803260761 Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809508 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe94e149320a4938216f3027cbe0604ed Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe94e149320a4938216f3027cbe0604ed;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=eae726c9592cbf633ada592803260761 To: sip:8850501@10.144.21.42;tag=as6739302b Call-ID: b034b607fe136cc347a3edea69432342@10.66.11.11 CSeq: 1605809508 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5044' G7-VOIPSERV*CLI> == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' G7-VOIPSERV*CLI> Really destroying SIP dialog 'b034b607fe136cc347a3edea69432342@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=6e175bdce4decb4383bc65eb99a3b6dc Contact: sip:10.66.11.11 Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637116 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK476c50ed20b8a646c139d1edefb0802b Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717492 1244717492 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32648 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32648 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32648 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK476c50ed20b8a646c139d1edefb0802b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=6e175bdce4decb4383bc65eb99a3b6dc To: sip:8850501@10.144.21.42 Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637116 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-2114 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK476c50ed20b8a646c139d1edefb0802b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=6e175bdce4decb4383bc65eb99a3b6dc To: sip:8850501@10.144.21.42;tag=as6e520797 Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637116 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-2114 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 13052 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK476c50ed20b8a646c139d1edefb0802b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=6e175bdce4decb4383bc65eb99a3b6dc To: sip:8850501@10.144.21.42;tag=as6e520797 Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637116 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13052 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6e520797 From: "Poste OP2 cmp 2" ;tag=6e175bdce4decb4383bc65eb99a3b6dc Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637116 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK510fefe3c8e8baac8b791b49110ff49c Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6e520797 From: sip:86235@10.66.11.11;tag=6e175bdce4decb4383bc65eb99a3b6dc Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637117 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8ae45afcc0d55d190c2e8dbc79b91dd Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8ae45afcc0d55d190c2e8dbc79b91dd;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=6e175bdce4decb4383bc65eb99a3b6dc To: sip:8850501@10.144.21.42;tag=as6e520797 Call-ID: 63cc28054a935dadbc270168b9ea1b81@10.66.11.11 CSeq: 1530637117 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-2114' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '63cc28054a935dadbc270168b9ea1b81@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=377cd6eb10ae687336a82b924678de66 Contact: sip:10.66.11.11 Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900501 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2f0059c4fdea1d26a1111824c47867a1 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717494 1244717494 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32664 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 G7-VOIPSERV*CLI> Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32664 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32664 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2f0059c4fdea1d26a1111824c47867a1;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=377cd6eb10ae687336a82b924678de66 To: sip:8850501@10.144.21.42 Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900501 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-8341 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2f0059c4fdea1d26a1111824c47867a1;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=377cd6eb10ae687336a82b924678de66 To: sip:8850501@10.144.21.42;tag=as475ab16d Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900501 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-8341 answered SIP/10.66.11.11-101d0ac8 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 12804 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2f0059c4fdea1d26a1111824c47867a1;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=377cd6eb10ae687336a82b924678de66 To: sip:8850501@10.144.21.42;tag=as475ab16d Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900501 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12804 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as475ab16d From: "Poste OP2 cmp 2" ;tag=377cd6eb10ae687336a82b924678de66 Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900501 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe3fca90f236cc1bb08a72cd349c5bab2 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as475ab16d From: sip:86235@10.66.11.11;tag=377cd6eb10ae687336a82b924678de66 Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900502 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2e2151195faced93696b1b12575bb9ea Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2e2151195faced93696b1b12575bb9ea;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=377cd6eb10ae687336a82b924678de66 To: sip:8850501@10.144.21.42;tag=as475ab16d Call-ID: 0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11 CSeq: 205900502 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-8341' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '0910d0ed3760b6bcaae5b9e51566b870@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=f4bca37d788517c3bcff7a64214ba62c Contact: sip:10.66.11.11 Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844299 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK26ec209f6d5dd5727233f203dc56cbfa Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717497 1244717497 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32680 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 448402726e2ac1e23307e16422525af4@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32680 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32680 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK26ec209f6d5dd5727233f203dc56cbfa;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=f4bca37d788517c3bcff7a64214ba62c To: sip:8850501@10.144.21.42 Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844299 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14058 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK26ec209f6d5dd5727233f203dc56cbfa;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=f4bca37d788517c3bcff7a64214ba62c To: sip:8850501@10.144.21.42;tag=as559dda80 Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844299 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14058 answered SIP/10.66.11.11-101d0ac8 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 14598 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK26ec209f6d5dd5727233f203dc56cbfa;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=f4bca37d788517c3bcff7a64214ba62c To: sip:8850501@10.144.21.42;tag=as559dda80 Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844299 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14598 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as559dda80 From: "Poste OP2 cmp 2" ;tag=f4bca37d788517c3bcff7a64214ba62c Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844299 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1f6420700bf20bc0fa9ff01e6e57d12b Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as559dda80 From: sip:86235@10.66.11.11;tag=f4bca37d788517c3bcff7a64214ba62c Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844300 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0cb3d7ce0c5fb995d1f8dcfc9c7cfd62 Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0cb3d7ce0c5fb995d1f8dcfc9c7cfd62;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=f4bca37d788517c3bcff7a64214ba62c To: sip:8850501@10.144.21.42;tag=as559dda80 Call-ID: 448402726e2ac1e23307e16422525af4@10.66.11.11 CSeq: 1844844300 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14058' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '448402726e2ac1e23307e16422525af4@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=ffa54c156870009bda21c3707ea99b49 Contact: sip:10.66.11.11 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675393 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd859616c1c3e77c24051be654f911eb6 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717502 1244717502 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32704 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - ead56bece40c38a8d46403e6392fc34a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32704 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32704 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd859616c1c3e77c24051be654f911eb6;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ffa54c156870009bda21c3707ea99b49 To: sip:8850501@10.144.21.42 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675393 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-2351 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd859616c1c3e77c24051be654f911eb6;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ffa54c156870009bda21c3707ea99b49 To: sip:8850501@10.144.21.42;tag=as40663ae1 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675393 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-2351 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 15220 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd859616c1c3e77c24051be654f911eb6;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ffa54c156870009bda21c3707ea99b49 To: sip:8850501@10.144.21.42;tag=as40663ae1 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675393 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 15220 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as40663ae1 From: "Poste OP2 cmp 2" ;tag=ffa54c156870009bda21c3707ea99b49 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675393 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKab7970949d8b5e560fb4c29a1a6effe2 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as40663ae1 From: sip:86235@10.66.11.11;tag=ffa54c156870009bda21c3707ea99b49 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675394 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKec6af708ae22e14a275365fc572c6725 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKec6af708ae22e14a275365fc572c6725;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=ffa54c156870009bda21c3707ea99b49 To: sip:8850501@10.144.21.42;tag=as40663ae1 Call-ID: ead56bece40c38a8d46403e6392fc34a@10.66.11.11 CSeq: 187675394 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-2351' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'ead56bece40c38a8d46403e6392fc34a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=c9b923743f3a3276c30d35420d7bf490 Contact: sip:10.66.11.11 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949139 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6091f61e64f02fdc9cccba0d232c9dc3 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717513 1244717513 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32728 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32728 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32728 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6091f61e64f02fdc9cccba0d232c9dc3;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c9b923743f3a3276c30d35420d7bf490 To: sip:8850501@10.144.21.42 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949139 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13377 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6091f61e64f02fdc9cccba0d232c9dc3;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c9b923743f3a3276c30d35420d7bf490 To: sip:8850501@10.144.21.42;tag=as4cbe2377 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949139 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13377 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 12818 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6091f61e64f02fdc9cccba0d232c9dc3;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c9b923743f3a3276c30d35420d7bf490 To: sip:8850501@10.144.21.42;tag=as4cbe2377 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949139 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12818 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4cbe2377 From: "Poste OP2 cmp 2" ;tag=c9b923743f3a3276c30d35420d7bf490 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949139 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK92228c0125e587eb3e7ca19e6e8cece1 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as4cbe2377 From: sip:86235@10.66.11.11;tag=c9b923743f3a3276c30d35420d7bf490 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949140 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKaef5a4f8e75998f2144a3d73cde31b85 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKaef5a4f8e75998f2144a3d73cde31b85;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=c9b923743f3a3276c30d35420d7bf490 To: sip:8850501@10.144.21.42;tag=as4cbe2377 Call-ID: 7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11 CSeq: 1830949140 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13377' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '7a3a1c6035591a6acf8e0829e5f7a2d5@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=22066f40c9a4f28430308d7a584960fe Contact: sip:10.66.11.11 Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279568 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbb8b53eaecb4226a528b470a4c473b45 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717516 1244717516 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32544 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32544 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32544 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbb8b53eaecb4226a528b470a4c473b45;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=22066f40c9a4f28430308d7a584960fe To: sip:8850501@10.144.21.42 Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-9855 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbb8b53eaecb4226a528b470a4c473b45;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=22066f40c9a4f28430308d7a584960fe To: sip:8850501@10.144.21.42;tag=as06a57373 Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-9855 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 11972 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbb8b53eaecb4226a528b470a4c473b45;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=22066f40c9a4f28430308d7a584960fe To: sip:8850501@10.144.21.42;tag=as06a57373 Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279568 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11972 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as06a57373 From: "Poste OP2 cmp 2" ;tag=22066f40c9a4f28430308d7a584960fe Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279568 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK854b19f505cac4f2e32d76a8788dc22a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as06a57373 From: sip:86235@10.66.11.11;tag=22066f40c9a4f28430308d7a584960fe Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279569 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbd7a92c50ab7139217633d038ea79d57 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbd7a92c50ab7139217633d038ea79d57;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=22066f40c9a4f28430308d7a584960fe To: sip:8850501@10.144.21.42;tag=as06a57373 Call-ID: 639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11 CSeq: 1643279569 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-9855' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog '639a2ff757c7f0cae54aa190a4f4c8cd@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 Contact: sip:10.66.11.11 Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128220 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK477a0563f7cd711904062260875f6dc0 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717519 1244717519 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32560 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - df42251b92c421048d73d38fafd0e105@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32560 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 G7-VOIPSERV*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32560 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK477a0563f7cd711904062260875f6dc0;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 To: sip:8850501@10.144.21.42 Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128220 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-6433 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK477a0563f7cd711904062260875f6dc0;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 To: sip:8850501@10.144.21.42;tag=as005b1cce Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128220 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-6433 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 16134 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK477a0563f7cd711904062260875f6dc0;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 To: sip:8850501@10.144.21.42;tag=as005b1cce Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128220 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16134 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as005b1cce From: "Poste OP2 cmp 2" ;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128220 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc13765f920c98dc3546bc4a1ed923ff0 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as005b1cce From: sip:86235@10.66.11.11;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128221 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKedddf4e74bb420f91d009c8cb53b02f6 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKedddf4e74bb420f91d009c8cb53b02f6;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=35ec8b0e2c31a6bf2fb225ef5dddb922 To: sip:8850501@10.144.21.42;tag=as005b1cce Call-ID: df42251b92c421048d73d38fafd0e105@10.66.11.11 CSeq: 372128221 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-6433' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'df42251b92c421048d73d38fafd0e105@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=ff388a726c6c2c3cec519df64ad12f50 Contact: sip:10.66.11.11 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845361 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK06dd78469ae0b09575f0b664e9218329 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717522 1244717522 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32568 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - a1c5507e81240e60645ac5d58215bd12@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32568 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32568 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK06dd78469ae0b09575f0b664e9218329;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ff388a726c6c2c3cec519df64ad12f50 To: sip:8850501@10.144.21.42 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845361 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-1675 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK06dd78469ae0b09575f0b664e9218329;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ff388a726c6c2c3cec519df64ad12f50 To: sip:8850501@10.144.21.42;tag=as363a1cf0 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845361 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-1675 answered SIP/10.66.11.11-101d5c48 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 14040 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK06dd78469ae0b09575f0b664e9218329;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ff388a726c6c2c3cec519df64ad12f50 To: sip:8850501@10.144.21.42;tag=as363a1cf0 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845361 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14040 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as363a1cf0 From: "Poste OP2 cmp 2" ;tag=ff388a726c6c2c3cec519df64ad12f50 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845361 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK05c47b479c1117069942c163c22f7f4f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as363a1cf0 From: sip:86235@10.66.11.11;tag=ff388a726c6c2c3cec519df64ad12f50 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845362 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfa7e0d5ad2a32184f75f92e9ab8b064e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfa7e0d5ad2a32184f75f92e9ab8b064e;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=ff388a726c6c2c3cec519df64ad12f50 To: sip:8850501@10.144.21.42;tag=as363a1cf0 Call-ID: a1c5507e81240e60645ac5d58215bd12@10.66.11.11 CSeq: 122845362 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-1675' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'a1c5507e81240e60645ac5d58215bd12@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=1b4c66638b42f3c2215de54a8769b1e1 Contact: sip:10.66.11.11 Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164976 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8731284ca84a3bd750c261987afa541b Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717526 1244717526 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32584 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 3d8082967c462cc39215be1562b9df50@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32584 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32584 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8731284ca84a3bd750c261987afa541b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=1b4c66638b42f3c2215de54a8769b1e1 To: sip:8850501@10.144.21.42 Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164976 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5648 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8731284ca84a3bd750c261987afa541b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=1b4c66638b42f3c2215de54a8769b1e1 To: sip:8850501@10.144.21.42;tag=as73ca468e Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164976 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5648 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19960 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8731284ca84a3bd750c261987afa541b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=1b4c66638b42f3c2215de54a8769b1e1 To: sip:8850501@10.144.21.42;tag=as73ca468e Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164976 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19960 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as73ca468e From: "Poste OP2 cmp 2" ;tag=1b4c66638b42f3c2215de54a8769b1e1 Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164976 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa04b5c5e2a339c6e225a91c56a23679e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as73ca468e From: sip:86235@10.66.11.11;tag=1b4c66638b42f3c2215de54a8769b1e1 Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164977 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe767b5acd40cf1c7a4694f7767afd284 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe767b5acd40cf1c7a4694f7767afd284;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=1b4c66638b42f3c2215de54a8769b1e1 To: sip:8850501@10.144.21.42;tag=as73ca468e Call-ID: 3d8082967c462cc39215be1562b9df50@10.66.11.11 CSeq: 1722164977 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5648' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '3d8082967c462cc39215be1562b9df50@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=01f11c617508b732541369183dceea12 Contact: sip:10.66.11.11 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386507 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb531530330c32d693c367a7982447f4e Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717529 1244717529 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32600 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32600 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) G7-VOIPSERV*CLI> Peer audio RTP is at port 10.144.27.22:32600 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb531530330c32d693c367a7982447f4e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=01f11c617508b732541369183dceea12 To: sip:8850501@10.144.21.42 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386507 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-12890 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb531530330c32d693c367a7982447f4e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=01f11c617508b732541369183dceea12 To: sip:8850501@10.144.21.42;tag=as77814400 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386507 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-12890 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 18566 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb531530330c32d693c367a7982447f4e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=01f11c617508b732541369183dceea12 To: sip:8850501@10.144.21.42;tag=as77814400 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386507 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 18566 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as77814400 From: "Poste OP2 cmp 2" ;tag=01f11c617508b732541369183dceea12 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386507 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2c5fa35bc93128bfcaa2f9ac0db16010 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as77814400 From: sip:86235@10.66.11.11;tag=01f11c617508b732541369183dceea12 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386508 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK70b01afe09ce34be1d51c83b07785ae3 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK70b01afe09ce34be1d51c83b07785ae3;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=01f11c617508b732541369183dceea12 To: sip:8850501@10.144.21.42;tag=as77814400 Call-ID: d5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11 CSeq: 1226386508 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-12890' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'd5aafb4cdb3d4febdcffb806b18f7ace@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=d77d559c246f365da0137f03bd342b05 Contact: sip:10.66.11.11 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273188 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb409baf1969fe45cc2528a5467c2658d Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717535 1244717535 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32616 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32616 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32616 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb409baf1969fe45cc2528a5467c2658d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=d77d559c246f365da0137f03bd342b05 To: sip:8850501@10.144.21.42 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273188 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5675 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb409baf1969fe45cc2528a5467c2658d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=d77d559c246f365da0137f03bd342b05 To: sip:8850501@10.144.21.42;tag=as0ef33c28 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273188 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5675 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19750 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb409baf1969fe45cc2528a5467c2658d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=d77d559c246f365da0137f03bd342b05 To: sip:8850501@10.144.21.42;tag=as0ef33c28 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273188 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19750 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0ef33c28 From: "Poste OP2 cmp 2" ;tag=d77d559c246f365da0137f03bd342b05 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273188 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9c517593df12e8fa68c9ea5b6538b9fb Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0ef33c28 From: sip:86235@10.66.11.11;tag=d77d559c246f365da0137f03bd342b05 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273189 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKee232053b6467899a570230e14db9ecd Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKee232053b6467899a570230e14db9ecd;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=d77d559c246f365da0137f03bd342b05 To: sip:8850501@10.144.21.42;tag=as0ef33c28 Call-ID: 3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11 CSeq: 1641273189 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5675' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '3bae22ddfcd25fc483fb10cf71e668ee@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=82d53bb45ab26602452fb2a311a62c63 Contact: sip:10.66.11.11 Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658106 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45752befbcffffefcb6321465cd9d264 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244717539 1244717539 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32632 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - e915b80772c00049101d302995d7b2da@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32632 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32632 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45752befbcffffefcb6321465cd9d264;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=82d53bb45ab26602452fb2a311a62c63 To: sip:8850501@10.144.21.42 Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-8064 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45752befbcffffefcb6321465cd9d264;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=82d53bb45ab26602452fb2a311a62c63 To: sip:8850501@10.144.21.42;tag=as31b2ac5d Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-8064 answered SIP/10.66.11.11-101d5c48 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 10804 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45752befbcffffefcb6321465cd9d264;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=82d53bb45ab26602452fb2a311a62c63 To: sip:8850501@10.144.21.42;tag=as31b2ac5d Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 10804 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as31b2ac5d From: "Poste OP2 cmp 2" ;tag=82d53bb45ab26602452fb2a311a62c63 Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658106 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK17c9c35b9a12d684e21fe34b7f4109ff Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as31b2ac5d From: sip:86235@10.66.11.11;tag=82d53bb45ab26602452fb2a311a62c63 Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658107 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4b7bb27ec9e8bfbaaf6ca87756a2b863 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4b7bb27ec9e8bfbaaf6ca87756a2b863;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=82d53bb45ab26602452fb2a311a62c63 To: sip:8850501@10.144.21.42;tag=as31b2ac5d Call-ID: e915b80772c00049101d302995d7b2da@10.66.11.11 CSeq: 234658107 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-8064' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'e915b80772c00049101d302995d7b2da@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> <-------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=9317057a69c83b4411284a4393628d5b Contact: sip:10.66.11.11 Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584563 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3f3da2e5aa045f3a5db5fd67e5b3b4 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718407 1244718407 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32656 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - e4900cd209b4e649c4058223a24156e9@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32656 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 G7-VOIPSERV*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32656 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3f3da2e5aa045f3a5db5fd67e5b3b4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9317057a69c83b4411284a4393628d5b To: sip:8850501@10.144.21.42 Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584563 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-3624 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3f3da2e5aa045f3a5db5fd67e5b3b4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9317057a69c83b4411284a4393628d5b To: sip:8850501@10.144.21.42;tag=as31625093 Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584563 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-3624 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 16468 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3f3da2e5aa045f3a5db5fd67e5b3b4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9317057a69c83b4411284a4393628d5b To: sip:8850501@10.144.21.42;tag=as31625093 Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584563 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16468 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as31625093 From: "Poste OP2 cmp 2" ;tag=9317057a69c83b4411284a4393628d5b Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584563 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb2491cdf8eda201f26ac4b184f1ff00f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as31625093 From: sip:86235@10.66.11.11;tag=9317057a69c83b4411284a4393628d5b Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584564 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe166fe61454e96e55a098ee269629978 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe166fe61454e96e55a098ee269629978;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=9317057a69c83b4411284a4393628d5b To: sip:8850501@10.144.21.42;tag=as31625093 Call-ID: e4900cd209b4e649c4058223a24156e9@10.66.11.11 CSeq: 2108584564 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-3624' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'e4900cd209b4e649c4058223a24156e9@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=0e1e1ee438bba588f2fe6a9c83db82f6 Contact: sip:10.66.11.11 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970597 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0a17a0d2b43fe700cf8c7d2fea060920 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718411 1244718411 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32672 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 83c749462364876d57775528c1966d95@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32672 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32672 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0a17a0d2b43fe700cf8c7d2fea060920;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0e1e1ee438bba588f2fe6a9c83db82f6 To: sip:8850501@10.144.21.42 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970597 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-1323 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0a17a0d2b43fe700cf8c7d2fea060920;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0e1e1ee438bba588f2fe6a9c83db82f6 To: sip:8850501@10.144.21.42;tag=as51c35dd8 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970597 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-1323 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 15242 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0a17a0d2b43fe700cf8c7d2fea060920;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0e1e1ee438bba588f2fe6a9c83db82f6 To: sip:8850501@10.144.21.42;tag=as51c35dd8 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970597 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 15242 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as51c35dd8 From: "Poste OP2 cmp 2" ;tag=0e1e1ee438bba588f2fe6a9c83db82f6 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970597 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0433c00a038874789d34ad6b1071fe30 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as51c35dd8 From: sip:86235@10.66.11.11;tag=0e1e1ee438bba588f2fe6a9c83db82f6 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970598 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK05535ad10f3d631360635ef7e8d4ba24 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK05535ad10f3d631360635ef7e8d4ba24;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=0e1e1ee438bba588f2fe6a9c83db82f6 To: sip:8850501@10.144.21.42;tag=as51c35dd8 Call-ID: 83c749462364876d57775528c1966d95@10.66.11.11 CSeq: 1638970598 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-1323' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '83c749462364876d57775528c1966d95@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=b3b43d50fab3d2ed8397db8bac486c92 Contact: sip:10.66.11.11 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638528 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK115f68e2627cfc03b39bd7359fe826db Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718499 1244718499 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32696 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - bec99f4af12214f60907a7152d635d81@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32696 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32696 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK115f68e2627cfc03b39bd7359fe826db;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b3b43d50fab3d2ed8397db8bac486c92 To: sip:8850501@10.144.21.42 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638528 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-8019 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK115f68e2627cfc03b39bd7359fe826db;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b3b43d50fab3d2ed8397db8bac486c92 To: sip:8850501@10.144.21.42;tag=as1c167a44 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638528 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-8019 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 18902 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK115f68e2627cfc03b39bd7359fe826db;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b3b43d50fab3d2ed8397db8bac486c92 To: sip:8850501@10.144.21.42;tag=as1c167a44 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638528 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 18902 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1c167a44 From: "Poste OP2 cmp 2" ;tag=b3b43d50fab3d2ed8397db8bac486c92 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638528 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf3bc5b7a7633d802ccd77acf8dc6feea Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1c167a44 From: sip:86235@10.66.11.11;tag=b3b43d50fab3d2ed8397db8bac486c92 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638529 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK274a185dff2bbd52f828e0a8f6cd537d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK274a185dff2bbd52f828e0a8f6cd537d;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=b3b43d50fab3d2ed8397db8bac486c92 To: sip:8850501@10.144.21.42;tag=as1c167a44 Call-ID: bec99f4af12214f60907a7152d635d81@10.66.11.11 CSeq: 371638529 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-8019' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'bec99f4af12214f60907a7152d635d81@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=2ed63869ffd90a94a744c07d7b2c4e77 Contact: sip:10.66.11.11 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978606 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK79e0d9085974ad9186254f63f88d3a83 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718507 1244718507 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32712 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32712 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32712 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK79e0d9085974ad9186254f63f88d3a83;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2ed63869ffd90a94a744c07d7b2c4e77 To: sip:8850501@10.144.21.42 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978606 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-11262 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK79e0d9085974ad9186254f63f88d3a83;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2ed63869ffd90a94a744c07d7b2c4e77 To: sip:8850501@10.144.21.42;tag=as61de6471 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978606 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-11262 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 17206 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK79e0d9085974ad9186254f63f88d3a83;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2ed63869ffd90a94a744c07d7b2c4e77 To: sip:8850501@10.144.21.42;tag=as61de6471 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978606 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17206 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as61de6471 From: "Poste OP2 cmp 2" ;tag=2ed63869ffd90a94a744c07d7b2c4e77 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978606 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3857c5e8c56af1c7d6c68c38cb873677 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as61de6471 From: sip:86235@10.66.11.11;tag=2ed63869ffd90a94a744c07d7b2c4e77 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978607 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20761170866f9d0d24309e0576131f2f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20761170866f9d0d24309e0576131f2f;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=2ed63869ffd90a94a744c07d7b2c4e77 To: sip:8850501@10.144.21.42;tag=as61de6471 Call-ID: fa2b89622b405c33d36bb0adc113c49a@10.66.11.11 CSeq: 640978607 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-11262' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'fa2b89622b405c33d36bb0adc113c49a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=686718c22e8f6c7a04fb79579ece9045 Contact: sip:10.66.11.11 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079717 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20c16af75e0d4e593643b9c0f5a9762b Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718512 1244718512 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32736 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - a09e388d73242657a750354548903c6d@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32736 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32736 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20c16af75e0d4e593643b9c0f5a9762b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=686718c22e8f6c7a04fb79579ece9045 To: sip:8850501@10.144.21.42 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079717 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13912 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20c16af75e0d4e593643b9c0f5a9762b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=686718c22e8f6c7a04fb79579ece9045 To: sip:8850501@10.144.21.42;tag=as6d267693 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079717 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13912 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19940 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK20c16af75e0d4e593643b9c0f5a9762b;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=686718c22e8f6c7a04fb79579ece9045 To: sip:8850501@10.144.21.42;tag=as6d267693 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079717 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19940 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6d267693 From: "Poste OP2 cmp 2" ;tag=686718c22e8f6c7a04fb79579ece9045 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079717 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7392cae0bb8804cdfc952ac61fac66e2 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6d267693 From: sip:86235@10.66.11.11;tag=686718c22e8f6c7a04fb79579ece9045 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079718 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa4d2873afa8a0c0a2db22f8bfacbfc62 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa4d2873afa8a0c0a2db22f8bfacbfc62;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=686718c22e8f6c7a04fb79579ece9045 To: sip:8850501@10.144.21.42;tag=as6d267693 Call-ID: a09e388d73242657a750354548903c6d@10.66.11.11 CSeq: 1054079718 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13912' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'a09e388d73242657a750354548903c6d@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=462d921af195575cb9f78144a8d5032d Contact: sip:10.66.11.11 Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728692 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45c159dbe171b02106f46854fbcec1c2 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718542 1244718542 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32552 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32552 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32552 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45c159dbe171b02106f46854fbcec1c2;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=462d921af195575cb9f78144a8d5032d To: sip:8850501@10.144.21.42 Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728692 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-2219 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45c159dbe171b02106f46854fbcec1c2;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=462d921af195575cb9f78144a8d5032d To: sip:8850501@10.144.21.42;tag=as7a4ebb13 Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728692 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-2219 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 11410 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK45c159dbe171b02106f46854fbcec1c2;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=462d921af195575cb9f78144a8d5032d To: sip:8850501@10.144.21.42;tag=as7a4ebb13 Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728692 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11410 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7a4ebb13 From: "Poste OP2 cmp 2" ;tag=462d921af195575cb9f78144a8d5032d Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728692 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdcb24c69b9d43401766e4c44d58fc17e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7a4ebb13 From: sip:86235@10.66.11.11;tag=462d921af195575cb9f78144a8d5032d Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728693 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16da6ae5b9d6aed534e6c45e2e986e5c Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16da6ae5b9d6aed534e6c45e2e986e5c;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=462d921af195575cb9f78144a8d5032d To: sip:8850501@10.144.21.42;tag=as7a4ebb13 Call-ID: ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11 CSeq: 379728693 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-2219' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'ccf5256fdd99429d98177dc41eee1cb2@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=26f7d274dd532f60cafbd59fd4a0a63a Contact: sip:10.66.11.11 Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361738 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK490065d2594ffc8afb4ac3a098f60960 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718554 1244718554 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32568 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32568 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) G7-VOIPSERV*CLI> Peer audio RTP is at port 10.144.27.22:32568 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK490065d2594ffc8afb4ac3a098f60960;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=26f7d274dd532f60cafbd59fd4a0a63a To: sip:8850501@10.144.21.42 Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361738 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-10209 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK490065d2594ffc8afb4ac3a098f60960;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=26f7d274dd532f60cafbd59fd4a0a63a To: sip:8850501@10.144.21.42;tag=as75369eff Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361738 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-10209 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14456 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK490065d2594ffc8afb4ac3a098f60960;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=26f7d274dd532f60cafbd59fd4a0a63a To: sip:8850501@10.144.21.42;tag=as75369eff Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361738 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14456 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as75369eff From: "Poste OP2 cmp 2" ;tag=26f7d274dd532f60cafbd59fd4a0a63a Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361738 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5b34f32294b8bafb7f2818e69455abba Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as75369eff From: sip:86235@10.66.11.11;tag=26f7d274dd532f60cafbd59fd4a0a63a Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361739 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8d03688504c88a48af962027127d115a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK8d03688504c88a48af962027127d115a;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=26f7d274dd532f60cafbd59fd4a0a63a To: sip:8850501@10.144.21.42;tag=as75369eff Call-ID: ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11 CSeq: 538361739 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-10209' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'ba9392216514ab0ba2537f317fdf6bf8@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=8b26e60e4ee911ee578396ca6f52ec11 Contact: sip:10.66.11.11 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886759 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81852066831ba3a5ffb200ef89f709d Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718558 1244718558 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32584 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 1d438a138132dfedb139b41dea341484@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32584 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32584 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81852066831ba3a5ffb200ef89f709d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=8b26e60e4ee911ee578396ca6f52ec11 To: sip:8850501@10.144.21.42 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886759 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) G7-VOIPSERV*CLI> -- Format for call is alaw G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-11340 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81852066831ba3a5ffb200ef89f709d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=8b26e60e4ee911ee578396ca6f52ec11 To: sip:8850501@10.144.21.42;tag=as0f533672 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886759 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-11340 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 16862 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb81852066831ba3a5ffb200ef89f709d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=8b26e60e4ee911ee578396ca6f52ec11 To: sip:8850501@10.144.21.42;tag=as0f533672 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886759 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16862 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0f533672 From: "Poste OP2 cmp 2" ;tag=8b26e60e4ee911ee578396ca6f52ec11 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886759 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9acd6342f134131ba4df620ba1309654 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0f533672 From: sip:86235@10.66.11.11;tag=8b26e60e4ee911ee578396ca6f52ec11 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886760 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3f9c300418663e4ad9444de23be0cc8c Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3f9c300418663e4ad9444de23be0cc8c;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=8b26e60e4ee911ee578396ca6f52ec11 To: sip:8850501@10.144.21.42;tag=as0f533672 Call-ID: 1d438a138132dfedb139b41dea341484@10.66.11.11 CSeq: 706886760 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-11340' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '1d438a138132dfedb139b41dea341484@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=9c22dc8410e6cb8958a29d9f8343f346 Contact: sip:10.66.11.11 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261244 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK866e2faa2ca4cb6a06ec9a26761d13ba Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718579 1244718579 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32600 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32600 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32600 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK866e2faa2ca4cb6a06ec9a26761d13ba;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9c22dc8410e6cb8958a29d9f8343f346 To: sip:8850501@10.144.21.42 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261244 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-445 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK866e2faa2ca4cb6a06ec9a26761d13ba;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9c22dc8410e6cb8958a29d9f8343f346 To: sip:8850501@10.144.21.42;tag=as51ff0570 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261244 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-445 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14408 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK866e2faa2ca4cb6a06ec9a26761d13ba;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9c22dc8410e6cb8958a29d9f8343f346 To: sip:8850501@10.144.21.42;tag=as51ff0570 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261244 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14408 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as51ff0570 From: "Poste OP2 cmp 2" ;tag=9c22dc8410e6cb8958a29d9f8343f346 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261244 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9d1e7538928303ed389227334b937170 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as51ff0570 From: sip:86235@10.66.11.11;tag=9c22dc8410e6cb8958a29d9f8343f346 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261245 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK67f8fbff30a79d8337cf06659b1dbb93 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK67f8fbff30a79d8337cf06659b1dbb93;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=9c22dc8410e6cb8958a29d9f8343f346 To: sip:8850501@10.144.21.42;tag=as51ff0570 Call-ID: 7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11 CSeq: 1550261245 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-445' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '7e578a5f9e150df97f390ff12e86e6aa@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=ed27e29b7af39c62f030659332d0e32d Contact: sip:10.66.11.11 Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960347 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK983104375423a0821dd6f454a868bbf4 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718582 1244718582 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32616 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32616 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32616 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK983104375423a0821dd6f454a868bbf4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ed27e29b7af39c62f030659332d0e32d To: sip:8850501@10.144.21.42 Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14826 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK983104375423a0821dd6f454a868bbf4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ed27e29b7af39c62f030659332d0e32d To: sip:8850501@10.144.21.42;tag=as79b2483d Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14826 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14510 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK983104375423a0821dd6f454a868bbf4;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ed27e29b7af39c62f030659332d0e32d To: sip:8850501@10.144.21.42;tag=as79b2483d Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14510 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as79b2483d From: "Poste OP2 cmp 2" ;tag=ed27e29b7af39c62f030659332d0e32d Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960347 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3d62531526271b8e7097226dd03ccb4e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as79b2483d From: sip:86235@10.66.11.11;tag=ed27e29b7af39c62f030659332d0e32d Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960348 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfbcfbf0d50c2c5f24ae546c3737a5c8c Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfbcfbf0d50c2c5f24ae546c3737a5c8c;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=ed27e29b7af39c62f030659332d0e32d To: sip:8850501@10.144.21.42;tag=as79b2483d Call-ID: 0fa25d3704001f554cb50bd13e67471c@10.66.11.11 CSeq: 382960348 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14826' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '0fa25d3704001f554cb50bd13e67471c@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=641da74b4bc1d1851aaa0161f57188eb Contact: sip:10.66.11.11 Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366154 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbd989034ec496e6a06c77a123591e99c Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718585 1244718585 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32632 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32632 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32632 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbd989034ec496e6a06c77a123591e99c;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=641da74b4bc1d1851aaa0161f57188eb To: sip:8850501@10.144.21.42 Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366154 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-3972 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbd989034ec496e6a06c77a123591e99c;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=641da74b4bc1d1851aaa0161f57188eb To: sip:8850501@10.144.21.42;tag=as79cc9fa7 Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366154 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-3972 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 12316 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbd989034ec496e6a06c77a123591e99c;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=641da74b4bc1d1851aaa0161f57188eb To: sip:8850501@10.144.21.42;tag=as79cc9fa7 Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366154 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12316 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as79cc9fa7 From: "Poste OP2 cmp 2" ;tag=641da74b4bc1d1851aaa0161f57188eb Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366154 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf26daadeeeb9267911a0440c0b6fc7e5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as79cc9fa7 From: sip:86235@10.66.11.11;tag=641da74b4bc1d1851aaa0161f57188eb Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366155 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7f8d1ef0bfe36dac830d6b684a229d46 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7f8d1ef0bfe36dac830d6b684a229d46;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=641da74b4bc1d1851aaa0161f57188eb To: sip:8850501@10.144.21.42;tag=as79cc9fa7 Call-ID: 2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11 CSeq: 1676366155 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-3972' G7-VOIPSERV*CLI> == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' G7-VOIPSERV*CLI> Really destroying SIP dialog '2bfc8ebf96eae676bdac3d313069d02c@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=43dff193af20c1b3f472ad233ca46a7c Contact: sip:10.66.11.11 Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960147 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3687837b6ee9ecccdcf3392903bf66 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718588 1244718588 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32656 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32656 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32656 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3687837b6ee9ecccdcf3392903bf66;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=43dff193af20c1b3f472ad233ca46a7c To: sip:8850501@10.144.21.42 Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960147 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-912 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3687837b6ee9ecccdcf3392903bf66;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=43dff193af20c1b3f472ad233ca46a7c To: sip:8850501@10.144.21.42;tag=as075bf23e Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960147 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-912 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 17530 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4a3687837b6ee9ecccdcf3392903bf66;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=43dff193af20c1b3f472ad233ca46a7c To: sip:8850501@10.144.21.42;tag=as075bf23e Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960147 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17530 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as075bf23e From: "Poste OP2 cmp 2" ;tag=43dff193af20c1b3f472ad233ca46a7c Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960147 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf83ddfdc7a5bbb8c3dd2459bbc48cdbe Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as075bf23e From: sip:86235@10.66.11.11;tag=43dff193af20c1b3f472ad233ca46a7c Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960148 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc0b78eecc7bc3dcacc16a43107de7a6f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc0b78eecc7bc3dcacc16a43107de7a6f;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=43dff193af20c1b3f472ad233ca46a7c To: sip:8850501@10.144.21.42;tag=as075bf23e Call-ID: 61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11 CSeq: 146960148 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-912' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '61b0647ce3fbc68bbe65383bfe2bbc53@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=ec07efdb29483c5c2cd1a9eba9a00be5 Contact: sip:10.66.11.11 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502364 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd44e9e819df07aceb127ae5c2cff9fff Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718591 1244718591 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32672 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32672 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32672 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd44e9e819df07aceb127ae5c2cff9fff;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ec07efdb29483c5c2cd1a9eba9a00be5 To: sip:8850501@10.144.21.42 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502364 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-2798 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd44e9e819df07aceb127ae5c2cff9fff;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ec07efdb29483c5c2cd1a9eba9a00be5 To: sip:8850501@10.144.21.42;tag=as002a77b3 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502364 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-2798 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13268 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKd44e9e819df07aceb127ae5c2cff9fff;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ec07efdb29483c5c2cd1a9eba9a00be5 To: sip:8850501@10.144.21.42;tag=as002a77b3 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502364 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13268 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as002a77b3 From: "Poste OP2 cmp 2" ;tag=ec07efdb29483c5c2cd1a9eba9a00be5 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502364 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf72785562e39047e078d725bdecd443a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as002a77b3 From: sip:86235@10.66.11.11;tag=ec07efdb29483c5c2cd1a9eba9a00be5 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502365 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3422208f3f42cf87e241d8dca2c68509 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3422208f3f42cf87e241d8dca2c68509;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=ec07efdb29483c5c2cd1a9eba9a00be5 To: sip:8850501@10.144.21.42;tag=as002a77b3 Call-ID: 0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11 CSeq: 161502365 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-2798' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '0c03268b778ddfe362ccbb0a9b74910a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=899dc8b584539824ba3287f37931cb5b Contact: sip:10.66.11.11 Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979433 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcfa4a286c988339b572ef5dc51eb8613 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718595 1244718595 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32696 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 G7-VOIPSERV*CLI> Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32696 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32696 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcfa4a286c988339b572ef5dc51eb8613;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=899dc8b584539824ba3287f37931cb5b To: sip:8850501@10.144.21.42 Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979433 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-15446 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcfa4a286c988339b572ef5dc51eb8613;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=899dc8b584539824ba3287f37931cb5b To: sip:8850501@10.144.21.42;tag=as3e137f39 Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979433 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-15446 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 16850 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcfa4a286c988339b572ef5dc51eb8613;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=899dc8b584539824ba3287f37931cb5b To: sip:8850501@10.144.21.42;tag=as3e137f39 Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979433 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16850 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as3e137f39 From: "Poste OP2 cmp 2" ;tag=899dc8b584539824ba3287f37931cb5b Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979433 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb8f667b519a23d4f78c1244043dbc35c Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as3e137f39 From: sip:86235@10.66.11.11;tag=899dc8b584539824ba3287f37931cb5b Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979434 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcc1cc31d02073c4eda79109f0386a42e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcc1cc31d02073c4eda79109f0386a42e;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=899dc8b584539824ba3287f37931cb5b To: sip:8850501@10.144.21.42;tag=as3e137f39 Call-ID: 4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11 CSeq: 1658979434 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-15446' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '4a296297c5f3a56a0ce00c2e8824cba6@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=b9af6adec19104cbe220908c039aaf57 Contact: sip:10.66.11.11 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323110 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0380c6fb839a977ff66067fd0f4f1278 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718600 1244718600 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32712 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32712 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32712 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0380c6fb839a977ff66067fd0f4f1278;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b9af6adec19104cbe220908c039aaf57 To: sip:8850501@10.144.21.42 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-4319 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0380c6fb839a977ff66067fd0f4f1278;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b9af6adec19104cbe220908c039aaf57 To: sip:8850501@10.144.21.42;tag=as0d7f4355 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-4319 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 11552 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0380c6fb839a977ff66067fd0f4f1278;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b9af6adec19104cbe220908c039aaf57 To: sip:8850501@10.144.21.42;tag=as0d7f4355 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11552 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0d7f4355 From: "Poste OP2 cmp 2" ;tag=b9af6adec19104cbe220908c039aaf57 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323110 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK01ca76fcac098087eab388741da22179 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0d7f4355 From: sip:86235@10.66.11.11;tag=b9af6adec19104cbe220908c039aaf57 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323111 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK21d780f1a4201f8f382aad61be91ab04 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK21d780f1a4201f8f382aad61be91ab04;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=b9af6adec19104cbe220908c039aaf57 To: sip:8850501@10.144.21.42;tag=as0d7f4355 Call-ID: cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11 CSeq: 1409323111 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-4319' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'cb3c42e7ee1ea9eed7b6bf670572f16c@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=7832accff74845ffcccff69f6c0a2caf Contact: sip:10.66.11.11 Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678334 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8285b958d860251c82d5d093d6e7a66 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718605 1244718605 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32736 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32736 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32736 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8285b958d860251c82d5d093d6e7a66;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7832accff74845ffcccff69f6c0a2caf To: sip:8850501@10.144.21.42 Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678334 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14728 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8285b958d860251c82d5d093d6e7a66;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7832accff74845ffcccff69f6c0a2caf To: sip:8850501@10.144.21.42;tag=as7eefed2d Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678334 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14728 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 13114 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe8285b958d860251c82d5d093d6e7a66;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7832accff74845ffcccff69f6c0a2caf To: sip:8850501@10.144.21.42;tag=as7eefed2d Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678334 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13114 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7eefed2d From: "Poste OP2 cmp 2" ;tag=7832accff74845ffcccff69f6c0a2caf Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678334 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe4c2c758bdaaac57860a1b1ee19a8998 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7eefed2d From: sip:86235@10.66.11.11;tag=7832accff74845ffcccff69f6c0a2caf Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678335 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK322758842419b576375a0283e4feecfa Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK322758842419b576375a0283e4feecfa;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=7832accff74845ffcccff69f6c0a2caf To: sip:8850501@10.144.21.42;tag=as7eefed2d Call-ID: cd2e05931f6e4539156b53de143e8bb6@10.66.11.11 CSeq: 1221678335 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14728' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'cd2e05931f6e4539156b53de143e8bb6@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=0416c6ca195ad84ab9b042d637b14b16 Contact: sip:10.66.11.11 Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345839 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9a22cfa0b5908a47094064e762bbb03f Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718608 1244718608 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32552 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32552 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32552 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9a22cfa0b5908a47094064e762bbb03f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0416c6ca195ad84ab9b042d637b14b16 To: sip:8850501@10.144.21.42 Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345839 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-9671 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9a22cfa0b5908a47094064e762bbb03f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0416c6ca195ad84ab9b042d637b14b16 To: sip:8850501@10.144.21.42;tag=as6ddbb81f Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345839 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-9671 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14504 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK9a22cfa0b5908a47094064e762bbb03f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=0416c6ca195ad84ab9b042d637b14b16 To: sip:8850501@10.144.21.42;tag=as6ddbb81f Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345839 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14504 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6ddbb81f From: "Poste OP2 cmp 2" ;tag=0416c6ca195ad84ab9b042d637b14b16 Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345839 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf47def54b71f7b1c7dde3033cf3dc814 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6ddbb81f From: sip:86235@10.66.11.11;tag=0416c6ca195ad84ab9b042d637b14b16 Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345840 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc6104f83920caf2eac15ac63d1f4704d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc6104f83920caf2eac15ac63d1f4704d;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=0416c6ca195ad84ab9b042d637b14b16 To: sip:8850501@10.144.21.42;tag=as6ddbb81f Call-ID: f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11 CSeq: 537345840 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-9671' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'f0cc258483a95cf0fa61bf9d654786b7@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=ad26796505a8a3b52271c98b3a6677cb Contact: sip:10.66.11.11 Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851800 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK90940372b022dec5e5ca65e335032703 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718616 1244718616 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32568 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 35e6154ced5123b48d99809d811ba391@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32568 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32568 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK90940372b022dec5e5ca65e335032703;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ad26796505a8a3b52271c98b3a6677cb To: sip:8850501@10.144.21.42 Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851800 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-8657 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK90940372b022dec5e5ca65e335032703;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ad26796505a8a3b52271c98b3a6677cb To: sip:8850501@10.144.21.42;tag=as60fad03b Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851800 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-8657 answered SIP/10.66.11.11-101d5c48 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 14086 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK90940372b022dec5e5ca65e335032703;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=ad26796505a8a3b52271c98b3a6677cb To: sip:8850501@10.144.21.42;tag=as60fad03b Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851800 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14086 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as60fad03b From: "Poste OP2 cmp 2" ;tag=ad26796505a8a3b52271c98b3a6677cb Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851800 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK678a582a871cd94f280f86728241134b Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as60fad03b From: sip:86235@10.66.11.11;tag=ad26796505a8a3b52271c98b3a6677cb Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851801 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK71343aba9643eeb618f607465dc0f2da Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK71343aba9643eeb618f607465dc0f2da;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=ad26796505a8a3b52271c98b3a6677cb To: sip:8850501@10.144.21.42;tag=as60fad03b Call-ID: 35e6154ced5123b48d99809d811ba391@10.66.11.11 CSeq: 1132851801 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-8657' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '35e6154ced5123b48d99809d811ba391@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=b0ad96fea292f431fe721f4d601a3170 Contact: sip:10.66.11.11 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038427 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK137ace071329ad2e93ead07730f203be Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718623 1244718623 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32584 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - bd5c4bc90f46688582c35863298d018c@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32584 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32584 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK137ace071329ad2e93ead07730f203be;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b0ad96fea292f431fe721f4d601a3170 To: sip:8850501@10.144.21.42 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038427 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13914 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK137ace071329ad2e93ead07730f203be;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b0ad96fea292f431fe721f4d601a3170 To: sip:8850501@10.144.21.42;tag=as5adf92a0 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038427 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13914 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 14550 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK137ace071329ad2e93ead07730f203be;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=b0ad96fea292f431fe721f4d601a3170 To: sip:8850501@10.144.21.42;tag=as5adf92a0 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038427 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14550 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as5adf92a0 From: "Poste OP2 cmp 2" ;tag=b0ad96fea292f431fe721f4d601a3170 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038427 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKea889253953337029ae6fdc00673b2e0 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as5adf92a0 From: sip:86235@10.66.11.11;tag=b0ad96fea292f431fe721f4d601a3170 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038428 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5ef6a272dbe7c7420336c18cdfbbf965 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5ef6a272dbe7c7420336c18cdfbbf965;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=b0ad96fea292f431fe721f4d601a3170 To: sip:8850501@10.144.21.42;tag=as5adf92a0 Call-ID: bd5c4bc90f46688582c35863298d018c@10.66.11.11 CSeq: 319038428 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13914' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'bd5c4bc90f46688582c35863298d018c@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 Contact: sip:10.66.11.11 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718625 1244718625 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32600 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32600 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32600 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-15226 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-15226 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 12174 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> CANCEL sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 To: sip:8850501@10.144.21.42 CSeq: 1774026484 CANCEL From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> Retransmitting #1 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as006f2084 From: sip:86235@10.66.11.11;tag=262b8bb382961ed2ffcc8250d85ecb25 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026485 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7ef12f433f4e9c670b978ef2a65dd7d9 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7ef12f433f4e9c670b978ef2a65dd7d9;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026485 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=6664ff62396fc0f490036e990117237a Contact: sip:10.66.11.11 Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403487 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1d5d9e4a914e55a86eef7b4f1b8d83c5 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718627 1244718627 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32616 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32616 G7-VOIPSERV*CLI> Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32616 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1d5d9e4a914e55a86eef7b4f1b8d83c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=6664ff62396fc0f490036e990117237a To: sip:8850501@10.144.21.42 Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403487 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-15226' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101ccc20", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13137 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1d5d9e4a914e55a86eef7b4f1b8d83c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=6664ff62396fc0f490036e990117237a To: sip:8850501@10.144.21.42;tag=as652e698a Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403487 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13137 answered SIP/10.66.11.11-101ccc20 Audio is at 10.144.21.42 port 11418 Retransmitting #2 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> Adding codec 0x4 (ulaw) to SDP G7-VOIPSERV*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1d5d9e4a914e55a86eef7b4f1b8d83c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=6664ff62396fc0f490036e990117237a To: sip:8850501@10.144.21.42;tag=as652e698a Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403487 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11418 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as652e698a From: "Poste OP2 cmp 2" ;tag=6664ff62396fc0f490036e990117237a Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403487 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0fd12c7df9442115e516b38c8d13d6bc Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as652e698a From: sip:86235@10.66.11.11;tag=6664ff62396fc0f490036e990117237a Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403488 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKaf789b197e498a90742cdb3c2d8f9465 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKaf789b197e498a90742cdb3c2d8f9465;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=6664ff62396fc0f490036e990117237a To: sip:8850501@10.144.21.42;tag=as652e698a Call-ID: 2d599bc0928996fa36b0a8ead173d47e@10.66.11.11 CSeq: 727403488 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13137' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101ccc20' Really destroying SIP dialog '2d599bc0928996fa36b0a8ead173d47e@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=27e367fed77682d06844ab9d808a9058 Contact: sip:10.66.11.11 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363633 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKabb5a166d26310a97c42cf94b3c68a6e Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718629 1244718629 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32632 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32632 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) G7-VOIPSERV*CLI> Peer audio RTP is at port 10.144.27.22:32632 G7-VOIPSERV*CLI> Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKabb5a166d26310a97c42cf94b3c68a6e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=27e367fed77682d06844ab9d808a9058 To: sip:8850501@10.144.21.42 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363633 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d71e0", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13694 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKabb5a166d26310a97c42cf94b3c68a6e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=27e367fed77682d06844ab9d808a9058 To: sip:8850501@10.144.21.42;tag=as0e14b7a2 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363633 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13694 answered SIP/10.66.11.11-101d71e0 Audio is at 10.144.21.42 port 16652 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKabb5a166d26310a97c42cf94b3c68a6e;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=27e367fed77682d06844ab9d808a9058 To: sip:8850501@10.144.21.42;tag=as0e14b7a2 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363633 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16652 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0e14b7a2 From: "Poste OP2 cmp 2" ;tag=27e367fed77682d06844ab9d808a9058 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363633 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf658f4454ced79e94ea063af46f69e1f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Retransmitting #3 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as0e14b7a2 From: sip:86235@10.66.11.11;tag=27e367fed77682d06844ab9d808a9058 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363634 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK66b7a86204d0423c949eed53b5525852 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK66b7a86204d0423c949eed53b5525852;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=27e367fed77682d06844ab9d808a9058 To: sip:8850501@10.144.21.42;tag=as0e14b7a2 Call-ID: c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11 CSeq: 874363634 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13694' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d71e0' Really destroying SIP dialog 'c9f523b05ea4d9dbc34f658c75be04a9@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=44da628aa647864cb929e6e27017a044 Contact: sip:10.66.11.11 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865996 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16df401e3145485d28284575775d6f5f Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244718631 1244718631 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32656 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 Found no matching peer or user for '10.66.11.11:10011' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32656 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32656 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16df401e3145485d28284575775d6f5f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=44da628aa647864cb929e6e27017a044 To: sip:8850501@10.144.21.42 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865996 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d71e0", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14682 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16df401e3145485d28284575775d6f5f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=44da628aa647864cb929e6e27017a044 To: sip:8850501@10.144.21.42;tag=as1e8e0161 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865996 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14682 answered SIP/10.66.11.11-101d71e0 Audio is at 10.144.21.42 port 12260 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16df401e3145485d28284575775d6f5f;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=44da628aa647864cb929e6e27017a044 To: sip:8850501@10.144.21.42;tag=as1e8e0161 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865996 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12260 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1e8e0161 From: "Poste OP2 cmp 2" ;tag=44da628aa647864cb929e6e27017a044 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865996 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK08f67cce64e57a2a517649165e125c6d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10011 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1e8e0161 From: sip:86235@10.66.11.11;tag=44da628aa647864cb929e6e27017a044 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865997 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa082487ab6678fa07b191ada9999ff5a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa082487ab6678fa07b191ada9999ff5a;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=44da628aa647864cb929e6e27017a044 To: sip:8850501@10.144.21.42;tag=as1e8e0161 Call-ID: ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11 CSeq: 1035865997 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14682' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d71e0' Really destroying SIP dialog 'ad3a467f629e19ca595b2fd5bbd6feff@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> Retransmitting #4 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> Retransmitting #5 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> Retransmitting #6 (no NAT) to 10.66.11.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK16678a7b717cd8a0dac0175d45510605;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=262b8bb382961ed2ffcc8250d85ecb25 To: sip:8850501@10.144.21.42;tag=as006f2084 Call-ID: d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 CSeq: 1774026484 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 12174 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- G7-VOIPSERV*CLI> [Nov 30 01:21:34] WARNING[464]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission d6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11 for seqno 1774026484 (Critical Response) Really destroying SIP dialog 'd6d2698a4a1b82b1e7ee5141e815558a@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10011 ---> <-------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=3eb200a9554ad34bc833a189e9b8c484 Contact: sip:10.66.11.11 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684890 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKacd2e0e1de709dc9804b7bbadcf178f2 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719341 1244719341 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32672 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32672 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32672 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKacd2e0e1de709dc9804b7bbadcf178f2;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=3eb200a9554ad34bc833a189e9b8c484 To: sip:8850501@10.144.21.42 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684890 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-11270 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKacd2e0e1de709dc9804b7bbadcf178f2;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=3eb200a9554ad34bc833a189e9b8c484 To: sip:8850501@10.144.21.42;tag=as1cb985c2 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684890 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-11270 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 19870 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKacd2e0e1de709dc9804b7bbadcf178f2;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=3eb200a9554ad34bc833a189e9b8c484 To: sip:8850501@10.144.21.42;tag=as1cb985c2 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684890 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 19870 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1cb985c2 From: "Poste OP2 cmp 2" ;tag=3eb200a9554ad34bc833a189e9b8c484 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684890 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK570d617f25f0c951783326624f009425 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1cb985c2 From: sip:86235@10.66.11.11;tag=3eb200a9554ad34bc833a189e9b8c484 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684891 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0faea4b10c5ac986bc708e0e50a742da Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0faea4b10c5ac986bc708e0e50a742da;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=3eb200a9554ad34bc833a189e9b8c484 To: sip:8850501@10.144.21.42;tag=as1cb985c2 Call-ID: 3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11 CSeq: 1956684891 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-11270' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '3086d19a423d5984d1acbd7cd8c4c860@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=c23c4f824ffd52302fcc7097ff4b4cdf Contact: sip:10.66.11.11 Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879575 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK849ce3dded4b1a0ba4904c2bb7c4ca28 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719352 1244719352 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32696 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32696 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32696 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK849ce3dded4b1a0ba4904c2bb7c4ca28;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c23c4f824ffd52302fcc7097ff4b4cdf To: sip:8850501@10.144.21.42 Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879575 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-14343 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK849ce3dded4b1a0ba4904c2bb7c4ca28;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c23c4f824ffd52302fcc7097ff4b4cdf To: sip:8850501@10.144.21.42;tag=as592bc072 Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879575 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-14343 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 17704 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK849ce3dded4b1a0ba4904c2bb7c4ca28;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=c23c4f824ffd52302fcc7097ff4b4cdf To: sip:8850501@10.144.21.42;tag=as592bc072 Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879575 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17704 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as592bc072 From: "Poste OP2 cmp 2" ;tag=c23c4f824ffd52302fcc7097ff4b4cdf Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879575 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf779ae87f99e44928e1f3ffa73c0a89b Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as592bc072 From: sip:86235@10.66.11.11;tag=c23c4f824ffd52302fcc7097ff4b4cdf Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879576 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb600152a35ca4f76d9ac1f589a4f3c8a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb600152a35ca4f76d9ac1f589a4f3c8a;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=c23c4f824ffd52302fcc7097ff4b4cdf To: sip:8850501@10.144.21.42;tag=as592bc072 Call-ID: b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11 CSeq: 1030879576 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-14343' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'b6bd60ed2bb026a091c2162becc55cb2@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b Contact: sip:10.66.11.11 Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225413 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4223bdcba410df37d066581e5ee8e7b3 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719359 1244719359 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32712 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - e109929dc489a8692192c3b31afbafcd@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32712 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32712 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4223bdcba410df37d066581e5ee8e7b3;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b To: sip:8850501@10.144.21.42 Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225413 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-15983 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4223bdcba410df37d066581e5ee8e7b3;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b To: sip:8850501@10.144.21.42;tag=as553af8d5 Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225413 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-15983 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 16372 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4223bdcba410df37d066581e5ee8e7b3;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b To: sip:8850501@10.144.21.42;tag=as553af8d5 Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225413 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16372 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as553af8d5 From: "Poste OP2 cmp 2" ;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225413 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7d80a12f2b0411cf40e39081c19f9b3d Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as553af8d5 From: sip:86235@10.66.11.11;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225414 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK25c445502a2d2152e32b65af39fd919e Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK25c445502a2d2152e32b65af39fd919e;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=1d2a5cde6a09b0f4d4bdaa551582cc7b To: sip:8850501@10.144.21.42;tag=as553af8d5 Call-ID: e109929dc489a8692192c3b31afbafcd@10.66.11.11 CSeq: 1568225414 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-15983' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'e109929dc489a8692192c3b31afbafcd@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=f6c008869779fe8859b77f5915c72deb Contact: sip:10.66.11.11 Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593564 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb69f350ef6d3021b53792b2a74c57784 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719362 1244719362 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32736 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b163d474ce942172af5e21bf217b3114@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32736 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) G7-VOIPSERV*CLI> Peer audio RTP is at port 10.144.27.22:32736 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb69f350ef6d3021b53792b2a74c57784;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=f6c008869779fe8859b77f5915c72deb To: sip:8850501@10.144.21.42 Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593564 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-10288 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb69f350ef6d3021b53792b2a74c57784;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=f6c008869779fe8859b77f5915c72deb To: sip:8850501@10.144.21.42;tag=as181e3907 Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593564 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-10288 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 11220 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb69f350ef6d3021b53792b2a74c57784;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=f6c008869779fe8859b77f5915c72deb To: sip:8850501@10.144.21.42;tag=as181e3907 Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593564 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11220 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as181e3907 From: "Poste OP2 cmp 2" ;tag=f6c008869779fe8859b77f5915c72deb Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593564 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKfb04a267962f13af59f4ebce2d2ea27f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as181e3907 From: sip:86235@10.66.11.11;tag=f6c008869779fe8859b77f5915c72deb Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593565 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK234b75b20f2db45b159dfb0c9e1824aa Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK234b75b20f2db45b159dfb0c9e1824aa;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=f6c008869779fe8859b77f5915c72deb To: sip:8850501@10.144.21.42;tag=as181e3907 Call-ID: b163d474ce942172af5e21bf217b3114@10.66.11.11 CSeq: 553593565 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-10288' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'b163d474ce942172af5e21bf217b3114@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=fc212adfbdc08d65766a40b85237f5fd Contact: sip:10.66.11.11 Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631991 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK680609b310b20aca6bb3b5902e6363c5 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719364 1244719364 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32552 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32552 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) G7-VOIPSERV*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32552 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK680609b310b20aca6bb3b5902e6363c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=fc212adfbdc08d65766a40b85237f5fd To: sip:8850501@10.144.21.42 Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631991 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d0ac8", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-5007 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK680609b310b20aca6bb3b5902e6363c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=fc212adfbdc08d65766a40b85237f5fd To: sip:8850501@10.144.21.42;tag=as47dbf83b Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631991 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-5007 answered SIP/10.66.11.11-101d0ac8 Audio is at 10.144.21.42 port 13576 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK680609b310b20aca6bb3b5902e6363c5;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=fc212adfbdc08d65766a40b85237f5fd To: sip:8850501@10.144.21.42;tag=as47dbf83b Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631991 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 13576 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as47dbf83b From: "Poste OP2 cmp 2" ;tag=fc212adfbdc08d65766a40b85237f5fd Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631991 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdc6038744832e748b597d69e1235b781 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as47dbf83b From: sip:86235@10.66.11.11;tag=fc212adfbdc08d65766a40b85237f5fd Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631992 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2f81d284f216c71c684aade190ab4ff1 Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK2f81d284f216c71c684aade190ab4ff1;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=fc212adfbdc08d65766a40b85237f5fd To: sip:8850501@10.144.21.42;tag=as47dbf83b Call-ID: f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11 CSeq: 872631992 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-5007' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d0ac8' Really destroying SIP dialog 'f0b2e4dd4f1310a4691fd840673178b1@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=632e4480966e162374f7159d16d8ad21 Contact: sip:10.66.11.11 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174826 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK59189a5ded3e5cea33db29a52c6e1889 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719371 1244719371 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32568 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32568 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32568 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK59189a5ded3e5cea33db29a52c6e1889;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=632e4480966e162374f7159d16d8ad21 To: sip:8850501@10.144.21.42 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174826 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-2666 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK59189a5ded3e5cea33db29a52c6e1889;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=632e4480966e162374f7159d16d8ad21 To: sip:8850501@10.144.21.42;tag=as32cc5c42 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174826 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-2666 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 11582 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK59189a5ded3e5cea33db29a52c6e1889;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=632e4480966e162374f7159d16d8ad21 To: sip:8850501@10.144.21.42;tag=as32cc5c42 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174826 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11582 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as32cc5c42 From: "Poste OP2 cmp 2" ;tag=632e4480966e162374f7159d16d8ad21 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174826 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK4ffc0dbb35c0af1f5046bf75d7963628 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as32cc5c42 From: sip:86235@10.66.11.11;tag=632e4480966e162374f7159d16d8ad21 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174827 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKacdbdc2a451f2478bfe95879c342803a Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKacdbdc2a451f2478bfe95879c342803a;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=632e4480966e162374f7159d16d8ad21 To: sip:8850501@10.144.21.42;tag=as32cc5c42 Call-ID: f4ab354dc52728e56ac01ba268eacd91@10.66.11.11 CSeq: 499174827 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-2666' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'f4ab354dc52728e56ac01ba268eacd91@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=9fc3528a4e99df6b943c1f5557973d05 Contact: sip:10.66.11.11 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924001 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7951a1dc3e5e134302e5fa6b6cb49147 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719373 1244719373 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32584 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32584 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32584 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7951a1dc3e5e134302e5fa6b6cb49147;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9fc3528a4e99df6b943c1f5557973d05 To: sip:8850501@10.144.21.42 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13058 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7951a1dc3e5e134302e5fa6b6cb49147;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9fc3528a4e99df6b943c1f5557973d05 To: sip:8850501@10.144.21.42;tag=as1d136fc9 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13058 answered SIP/10.66.11.11-101d5c48 G7-VOIPSERV*CLI> Audio is at 10.144.21.42 port 14784 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7951a1dc3e5e134302e5fa6b6cb49147;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=9fc3528a4e99df6b943c1f5557973d05 To: sip:8850501@10.144.21.42;tag=as1d136fc9 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 14784 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1d136fc9 From: "Poste OP2 cmp 2" ;tag=9fc3528a4e99df6b943c1f5557973d05 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924001 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf81e9c1becd11be9bb0ef5ccbb7a25c9 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as1d136fc9 From: sip:86235@10.66.11.11;tag=9fc3528a4e99df6b943c1f5557973d05 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924002 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa67b6ad16d7149095f658925a841d8c3 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKa67b6ad16d7149095f658925a841d8c3;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=9fc3528a4e99df6b943c1f5557973d05 To: sip:8850501@10.144.21.42;tag=as1d136fc9 Call-ID: e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11 CSeq: 1907924002 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13058' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'e3a30a15f9889b0ab7645f611888c9ac@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=4e120d3ff11a0cdb1520c7521cda33f9 Contact: sip:10.66.11.11 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491150 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1cc3286b3b86c0f37ec99d749bdf320d Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719381 1244719381 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32600 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32600 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32600 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1cc3286b3b86c0f37ec99d749bdf320d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=4e120d3ff11a0cdb1520c7521cda33f9 To: sip:8850501@10.144.21.42 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491150 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-10726 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1cc3286b3b86c0f37ec99d749bdf320d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=4e120d3ff11a0cdb1520c7521cda33f9 To: sip:8850501@10.144.21.42;tag=as6e5dfae6 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491150 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-10726 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 11046 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1cc3286b3b86c0f37ec99d749bdf320d;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=4e120d3ff11a0cdb1520c7521cda33f9 To: sip:8850501@10.144.21.42;tag=as6e5dfae6 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491150 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11046 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6e5dfae6 From: "Poste OP2 cmp 2" ;tag=4e120d3ff11a0cdb1520c7521cda33f9 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491150 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcd61391077ecc0bf3017403dd895e9ef Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as6e5dfae6 From: sip:86235@10.66.11.11;tag=4e120d3ff11a0cdb1520c7521cda33f9 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491151 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7e5413fa31b654b4a855483df69bc063 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK7e5413fa31b654b4a855483df69bc063;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=4e120d3ff11a0cdb1520c7521cda33f9 To: sip:8850501@10.144.21.42;tag=as6e5dfae6 Call-ID: 7039d5f28552cf92799223d154a4d7a9@10.66.11.11 CSeq: 828491151 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-10726' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '7039d5f28552cf92799223d154a4d7a9@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=afffdecb90f13f85f85ada1926d737ac Contact: sip:10.66.11.11 Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097895 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK375ee6e7b0b43ae7c782c8aaa716b44c Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719385 1244719385 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32616 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32616 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32616 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK375ee6e7b0b43ae7c782c8aaa716b44c;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=afffdecb90f13f85f85ada1926d737ac To: sip:8850501@10.144.21.42 Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097895 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-11066 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK375ee6e7b0b43ae7c782c8aaa716b44c;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=afffdecb90f13f85f85ada1926d737ac To: sip:8850501@10.144.21.42;tag=as74baf33a Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097895 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-11066 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 18394 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK375ee6e7b0b43ae7c782c8aaa716b44c;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=afffdecb90f13f85f85ada1926d737ac To: sip:8850501@10.144.21.42;tag=as74baf33a Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097895 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 18394 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as74baf33a From: "Poste OP2 cmp 2" ;tag=afffdecb90f13f85f85ada1926d737ac Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097895 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK0940d3758563cbf39ea23f4aaa9c3967 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as74baf33a From: sip:86235@10.66.11.11;tag=afffdecb90f13f85f85ada1926d737ac Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097896 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbfb54b9ae8d8b0ad26c3c15f42d8a336 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKbfb54b9ae8d8b0ad26c3c15f42d8a336;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=afffdecb90f13f85f85ada1926d737ac To: sip:8850501@10.144.21.42;tag=as74baf33a Call-ID: ebd840200be9378cd3eb178f3ff5420d@10.66.11.11 CSeq: 1768097896 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-11066' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'ebd840200be9378cd3eb178f3ff5420d@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=7b7d4cb5c6b04766da59f5ecfa8bbead Contact: sip:10.66.11.11 Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496753 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1df97c4d2e79e5a458ce39d5a4e73fee Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719388 1244719388 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32632 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - f94cf738fc769604d21553d445c47240@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32632 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32632 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1df97c4d2e79e5a458ce39d5a4e73fee;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7b7d4cb5c6b04766da59f5ecfa8bbead To: sip:8850501@10.144.21.42 Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496753 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-4737 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1df97c4d2e79e5a458ce39d5a4e73fee;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7b7d4cb5c6b04766da59f5ecfa8bbead To: sip:8850501@10.144.21.42;tag=as7c3b1a6e Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496753 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-4737 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 17344 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK1df97c4d2e79e5a458ce39d5a4e73fee;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=7b7d4cb5c6b04766da59f5ecfa8bbead To: sip:8850501@10.144.21.42;tag=as7c3b1a6e Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496753 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17344 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7c3b1a6e From: "Poste OP2 cmp 2" ;tag=7b7d4cb5c6b04766da59f5ecfa8bbead Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496753 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK99954596adc6981f52780ec6cc677cc9 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as7c3b1a6e From: sip:86235@10.66.11.11;tag=7b7d4cb5c6b04766da59f5ecfa8bbead Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496754 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf2b426390cd607a5d44356b7163bd434 Max-Forwards: 70 Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKf2b426390cd607a5d44356b7163bd434;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=7b7d4cb5c6b04766da59f5ecfa8bbead To: sip:8850501@10.144.21.42;tag=as7c3b1a6e Call-ID: f94cf738fc769604d21553d445c47240@10.66.11.11 CSeq: 327496754 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-4737' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'f94cf738fc769604d21553d445c47240@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=aa8b724c137f17838f5fd05fbe7076e9 Contact: sip:10.66.11.11 Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170754 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK714b455387cd4d6739404da101209126 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719391 1244719391 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32656 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 670653627c3129e3c8dfd906a08d743d@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32656 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32656 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK714b455387cd4d6739404da101209126;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=aa8b724c137f17838f5fd05fbe7076e9 To: sip:8850501@10.144.21.42 Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170754 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-6087 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK714b455387cd4d6739404da101209126;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=aa8b724c137f17838f5fd05fbe7076e9 To: sip:8850501@10.144.21.42;tag=as23b0985d Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170754 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-6087 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 16484 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK714b455387cd4d6739404da101209126;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=aa8b724c137f17838f5fd05fbe7076e9 To: sip:8850501@10.144.21.42;tag=as23b0985d Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170754 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 16484 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as23b0985d From: "Poste OP2 cmp 2" ;tag=aa8b724c137f17838f5fd05fbe7076e9 Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170754 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK3e79c9f578075d01a81d7d9dfd235992 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as23b0985d From: sip:86235@10.66.11.11;tag=aa8b724c137f17838f5fd05fbe7076e9 Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170755 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe326755245f382aa4742627a9baf434c Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKe326755245f382aa4742627a9baf434c;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=aa8b724c137f17838f5fd05fbe7076e9 To: sip:8850501@10.144.21.42;tag=as23b0985d Call-ID: 670653627c3129e3c8dfd906a08d743d@10.66.11.11 CSeq: 193170755 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-6087' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog '670653627c3129e3c8dfd906a08d743d@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=68e61ded2bf5b51fbca441df7b15b487 Contact: sip:10.66.11.11 Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763852 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5e5e61e94f6dfaeeef1cd30ef2f3ffeb Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719395 1244719395 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32672 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - b2f168db9fd4717595f134da3eed7d83@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32672 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32672 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5e5e61e94f6dfaeeef1cd30ef2f3ffeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=68e61ded2bf5b51fbca441df7b15b487 To: sip:8850501@10.144.21.42 Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763852 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-6649 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5e5e61e94f6dfaeeef1cd30ef2f3ffeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=68e61ded2bf5b51fbca441df7b15b487 To: sip:8850501@10.144.21.42;tag=as11046a3b Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763852 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-6649 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 11736 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK5e5e61e94f6dfaeeef1cd30ef2f3ffeb;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=68e61ded2bf5b51fbca441df7b15b487 To: sip:8850501@10.144.21.42;tag=as11046a3b Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763852 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 11736 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as11046a3b From: "Poste OP2 cmp 2" ;tag=68e61ded2bf5b51fbca441df7b15b487 Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763852 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK017212fdc521d9538932e0b2d050c296 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as11046a3b From: sip:86235@10.66.11.11;tag=68e61ded2bf5b51fbca441df7b15b487 Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763853 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc8b60665ab58ba03c4c01fd9a49eb1fb Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKc8b60665ab58ba03c4c01fd9a49eb1fb;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=68e61ded2bf5b51fbca441df7b15b487 To: sip:8850501@10.144.21.42;tag=as11046a3b Call-ID: b2f168db9fd4717595f134da3eed7d83@10.66.11.11 CSeq: 1128763853 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-6649' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Really destroying SIP dialog 'b2f168db9fd4717595f134da3eed7d83@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc Contact: sip:10.66.11.11 Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281712 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb1b3e1aa2e343ee90f3b87534529f4e9 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719398 1244719398 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32696 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- G7-VOIPSERV*CLI> Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32696 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32696 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb1b3e1aa2e343ee90f3b87534529f4e9;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc To: sip:8850501@10.144.21.42 Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d5c48", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-7971 is ringing G7-VOIPSERV*CLI> <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb1b3e1aa2e343ee90f3b87534529f4e9;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc To: sip:8850501@10.144.21.42;tag=as16bf6d01 Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-7971 answered SIP/10.66.11.11-101d5c48 Audio is at 10.144.21.42 port 17340 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKb1b3e1aa2e343ee90f3b87534529f4e9;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc To: sip:8850501@10.144.21.42;tag=as16bf6d01 Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17340 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as16bf6d01 From: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281712 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK6ddc60c4358fc764ee21f606934eea6f Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as16bf6d01 From: sip:86235@10.66.11.11;tag=2fd2a2b0a6a809730f01cd314e4880fc Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281713 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcb61a292ea99c506e391c95d49fe0806 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Nov 30 01:34:07] ERROR[464]: chan_sip.c:15553 sipsock_read: We could NOT get the channel lock for SIP/10.66.11.11-101d5c48! [Nov 30 01:34:07] ERROR[464]: chan_sip.c:15554 sipsock_read: SIP transaction failed: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKcb61a292ea99c506e391c95d49fe0806;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=2fd2a2b0a6a809730f01cd314e4880fc To: sip:8850501@10.144.21.42;tag=as16bf6d01 Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 873281713 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> INVITE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "Poste OP2 cmp 2" Content-Type: application/sdp To: sip:8850501@10.144.21.42 From: "Poste OP2 cmp 2" ;tag=95c0016dc847ff6c40ea08b84ae15e1c Contact: sip:10.66.11.11 Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378621 INVITE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK502d91b22c7251004f18314670b2ca46 Max-Forwards: 70 Content-Length: 315 v=0 o=OXE 1244719401 1244719401 IN IP4 10.66.11.11 s=abs c=IN IP4 10.144.27.22 t=0 0 m=audio 32712 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) Using INVITE request as basis request - 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 G7-VOIPSERV*CLI> Found no matching peer or user for '10.66.11.11:10012' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 10.144.27.22:32712 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.144.27.22:32712 Looking for 8850501 in default (domain 10.144.21.42) list_route: hop: <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK502d91b22c7251004f18314670b2ca46;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=95c0016dc847ff6c40ea08b84ae15e1c To: sip:8850501@10.144.21.42 Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378621 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Executing [8850501@default:1] Dial("SIP/10.66.11.11-101d34c0", "IAX2/DSP-IAX-05/0501") in new stack G7-VOIPSERV*CLI> -- Called DSP-IAX-05/0501 G7-VOIPSERV*CLI> -- Call accepted by 10.144.21.43 (format alaw) -- Format for call is alaw -- IAX2/DSP-IAX-05-13843 is ringing <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK502d91b22c7251004f18314670b2ca46;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=95c0016dc847ff6c40ea08b84ae15e1c To: sip:8850501@10.144.21.42;tag=as333ff599 Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378621 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- IAX2/DSP-IAX-05-13843 answered SIP/10.66.11.11-101d34c0 Audio is at 10.144.21.42 port 17166 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bK502d91b22c7251004f18314670b2ca46;received=10.66.11.11 From: "Poste OP2 cmp 2" ;tag=95c0016dc847ff6c40ea08b84ae15e1c To: sip:8850501@10.144.21.42;tag=as333ff599 Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378621 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 408 408 IN IP4 10.144.21.42 s=session c=IN IP4 10.144.21.42 t=0 0 m=audio 17166 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> ACK sip:8850501@10.144.21.42 SIP/2.0 Contact: sip:10.66.11.11 User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as333ff599 From: "Poste OP2 cmp 2" ;tag=95c0016dc847ff6c40ea08b84ae15e1c Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378621 ACK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKedd25ed1cc62b794c4cc128b1975ed4b Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.66.11.11:10012 ---> BYE sip:8850501@10.144.21.42 SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 To: sip:8850501@10.144.21.42;tag=as333ff599 From: sip:86235@10.66.11.11;tag=95c0016dc847ff6c40ea08b84ae15e1c Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378622 BYE Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdbf25b558981a2d728b5d309a72b6100 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.66.11.11 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.66.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.66.11.11;branch=z9hG4bKdbf25b558981a2d728b5d309a72b6100;received=10.66.11.11 From: sip:86235@10.66.11.11;tag=95c0016dc847ff6c40ea08b84ae15e1c To: sip:8850501@10.144.21.42;tag=as333ff599 Call-ID: 8edb21774de1c69fc697ea0aea89618f@10.66.11.11 CSeq: 1035378622 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-13843' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d34c0' Really destroying SIP dialog '8edb21774de1c69fc697ea0aea89618f@10.66.11.11' Method: BYE G7-VOIPSERV*CLI> -- Hungup 'IAX2/DSP-IAX-05-7971' == Spawn extension (default, 8850501, 1) exited non-zero on 'SIP/10.66.11.11-101d5c48' Scheduling destruction of SIP dialog '5bc212440f88e8bf49494b2c637965ed@10.66.11.11' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.66.11.11, port 5060 Reliably Transmitting (no NAT) to 10.66.11.11:5060: BYE sip:10.66.11.11 SIP/2.0 Via: SIP/2.0/UDP 10.144.21.42:5060;branch=z9hG4bK39451437;rport From: sip:8850501@10.144.21.42;tag=as16bf6d01 To: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- G7-VOIPSERV*CLI> <--- SIP read from 10.66.11.11:10012 ---> SIP/2.0 404 Not Found To: "Poste OP2 cmp 2" ;tag=2fd2a2b0a6a809730f01cd314e4880fc From: sip:8850501@10.144.21.42;tag=as16bf6d01 Call-ID: 5bc212440f88e8bf49494b2c637965ed@10.66.11.11 CSeq: 102 BYE Via: SIP/2.0/UDP 10.144.21.42:5060;branch=z9hG4bK39451437;rport Content-Length: 0 <-------------> G7-VOIPSERV*CLI> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived -- Incoming call: G