[Apr 28 13:39:24] <--- SIP read from UDP:10.251.18.235:5060 ---> INVITE sip:2249@asterisk SIP/2.0 Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK2036155347 From: "Unidata" ;tag=327081027 To: Supported: replaces, 100rel, timer Call-ID: 2138401834@10.251.18.235 CSeq: 20 INVITE Session-Expires: 600 Contact: Max-Forwards: 70 User-Agent: WPU-7700-v2.8.0/0 Expires: 180 Content-Type: application/sdp Content-Length: 275 v=0 =89019 1821135208 1918030179 IN IP4 10.251.18.235 s=A_converstion c=IN IP4 10.251.18.235 t=0 0 m=audio 9000 RTP/AVP 8 18 0 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Apr 28 13:39:24] --- (14 headers 12 lines) --- [Apr 28 13:39:24] Sending to 10.251.18.235:5060 (no NAT) [Apr 28 13:39:24] Using INVITE request as basis request - 2138401834@10.251.18.235 [Apr 28 13:39:24] Found peer '89019' for '89019' from 10.251.18.235:5060 [Apr 28 13:39:24] <--- Reliably Transmitting (no NAT) to 10.251.18.235:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK2036155347;received=10.251.18.235 From: "Unidata" ;tag=327081027 To: ;tag=as586edbd4 Call-ID: 2138401834@10.251.18.235 CSeq: 20 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="135bf924" Content-Length: 0 <------------> [Apr 28 13:39:24] Scheduling destruction of SIP dialog '2138401834@10.251.18.235' in 40576 ms (Method: INVITE) [Apr 28 13:39:24] <--- SIP read from UDP:10.251.18.235:5060 ---> ACK sip:2249@asterisk SIP/2.0 Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK2036155347 From: "Unidata" ;tag=327081027 To: ;tag=as586edbd4 Call-ID: 2138401834@10.251.18.235 CSeq: 20 ACK Content-Length: 0 <-------------> [Apr 28 13:39:24] --- (7 headers 0 lines) --- [Apr 28 13:39:24] <--- SIP read from UDP:10.251.18.235:5060 ---> INVITE sip:2249@asterisk SIP/2.0 Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK237231552 From: "Unidata" ;tag=327081027 To: Supported: replaces, 100rel, timer Call-ID: 2138401834@10.251.18.235 CSeq: 21 INVITE Session-Expires: 600 Contact: Authorization: Digest username="89019", realm="asterisk", nonce="135bf924", uri="sip:2249@asterisk", response="d08e7810577b9568872ad8623c4eb5d2", algorithm=MD5 Max-Forwards: 70 User-Agent: WPU-7700-v2.8.0/0 Expires: 180 Content-Type: application/sdp Content-Length: 275 v=0 o=89019 1821135208 1918030179 IN IP4 10.251.18.235 s=A_converstion c=IN IP4 10.251.18.235 t=0 0 m=audio 9000 RTP/AVP 8 18 0 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Apr 28 13:39:24] --- (15 headers 12 lines) --- [Apr 28 13:39:24] Sending to 10.251.18.235:5060 (no NAT) [Apr 28 13:39:24] Using INVITE request as basis request - 2138401834@10.251.18.235 [Apr 28 13:39:24] Found peer '89019' for '89019' from 10.251.18.235:5060 [Apr 28 13:39:24] == Using SIP RTP TOS bits 184 [Apr 28 13:39:24] == Using SIP RTP CoS mark 5 [Apr 28 13:39:24] Found RTP audio format 8 [Apr 28 13:39:24] Found RTP audio format 18 [Apr 28 13:39:24] Found RTP audio format 0 [Apr 28 13:39:24] Found RTP audio format 101 [Apr 28 13:39:24] Found audio description format PCMA for ID 8 [Apr 28 13:39:24] Found audio description format G729 for ID 18 [Apr 28 13:39:24] Found audio description format PCMU for ID 0 [Apr 28 13:39:24] Found audio description format telephone-event for ID 101 [Apr 28 13:39:24] Capabilities: us - 0x8 (alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Apr 28 13:39:24] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 28 13:39:24] Peer audio RTP is at port 10.251.18.235:9000 [Apr 28 13:39:24] Looking for 2249 in dialout-nopin (domain asterisk) [Apr 28 13:39:24] list_route: hop: [Apr 28 13:39:24] <--- Transmitting (no NAT) to 10.251.18.235:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK237231552;received=10.251.18.235 From: "Unidata" ;tag=327081027 To: Call-ID: 2138401834@10.251.18.235 CSeq: 21 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [Apr 28 13:39:24] -- Executing [2249@dialout-nopin:1] Dial("SIP/89019-00000000", "sip/vs10/2249") in new stack [Apr 28 13:39:24] == Using SIP RTP TOS bits 184 [Apr 28 13:39:24] == Using SIP RTP CoS mark 5 [Apr 28 13:39:24] Audio is at 5060 [Apr 28 13:39:24] Adding codec 0x8 (alaw) to SDP [Apr 28 13:39:24] Reliably Transmitting (no NAT) to 10.0.2.200:5060: INVITE sip:2249@10.0.2.200;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK6db599df Max-Forwards: 70 From: "Unidata" ;tag=as462aa110 To: Contact: Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Thu, 28 Apr 2011 11:39:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "Unidata" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 197 v=0 o=root 55210502 55210502 IN IP4 10.0.2.22 s=Asterisk PBX 1.8.3.2 c=IN IP4 10.0.2.22 t=0 0 m=audio 13628 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Apr 28 13:39:24] -- Called vs10/2249 [Apr 28 13:39:24] <--- SIP read from UDP:10.0.2.200:5060 ---> SIP/2.0 100 Trying From: "Unidata";tag=as462aa110 To: Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK6db599df Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> [Apr 28 13:39:24] --- (11 headers 0 lines) --- [Apr 28 13:39:24] <--- SIP read from UDP:10.0.2.200:5060 ---> SIP/2.0 180 Ringing From: "Unidata";tag=as462aa110 To: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK6db599df Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> [Apr 28 13:39:24] --- (11 headers 0 lines) --- [Apr 28 13:39:24] -- SIP/vs10-00000001 is ringing [Apr 28 13:39:24] <--- Transmitting (no NAT) to 10.251.18.235:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK237231552;received=10.251.18.235 From: "Unidata" ;tag=327081027 To: ;tag=as69233e59 Call-ID: 2138401834@10.251.18.235 CSeq: 21 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [Apr 28 13:39:27] <--- SIP read from UDP:10.0.2.200:5060 ---> SIP/2.0 200 OK From: "Unidata";tag=as462aa110 To: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK6db599df Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 P-Asserted-Identity: "Baldus Stefan" Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 210 v=0 o=- 30355 1 IN IP4 10.0.2.200 s=- t=0 0 m=audio 5200 RTP/AVP 8 101 111 c=IN IP4 10.0.23.188 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=sendrecv <-------------> [Apr 28 13:39:27] --- (14 headers 11 lines) --- [Apr 28 13:39:27] Found RTP audio format 8 [Apr 28 13:39:27] Found RTP audio format 101 [Apr 28 13:39:27] Found RTP audio format 111 [Apr 28 13:39:27] Found audio description format telephone-event for ID 101 [Apr 28 13:39:27] Found audio description format X-nt-inforeq for ID 111 [Apr 28 13:39:27] Capabilities: us - 0x8 (alaw), peer - audio=0x808 (alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Apr 28 13:39:27] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [Apr 28 13:39:27] Peer audio RTP is at port 10.0.23.188:5200 [Apr 28 13:39:27] list_route: hop: [Apr 28 13:39:27] Transmitting (no NAT) to 10.0.2.200:5060: ACK sip:2249;phone-context=UnknownUnknown@vs10.sip:5060;maddr=10.0.2.200;transport=udp;user=phone SIP/2.0 ia: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK0f55f605 Max-Forwards: 70 From: "Unidata" ;tag=as462aa110 To: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 Contact: Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- [Apr 28 13:39:27] -- SIP/vs10-00000001 answered SIP/89019-00000000 [Apr 28 13:39:27] Audio is at 5060 [Apr 28 13:39:27] Adding codec 0x8 (alaw) to SDP [Apr 28 13:39:27] Adding non-codec 0x1 (telephone-event) to SDP [Apr 28 13:39:27] <--- Reliably Transmitting (no NAT) to 10.251.18.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK237231552;received=10.251.18.235 From: "Unidata" ;tag=327081027 To: ;tag=as69233e59 Call-ID: 2138401834@10.251.18.235 CSeq: 21 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 255 v=0 o=root 386671502 386671502 IN IP4 10.0.2.22 s=Asterisk PBX 1.8.3.2 c=IN IP4 10.0.2.22 t=0 0 m=audio 15978 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 28 13:39:27] <--- SIP read from UDP:10.251.18.235:5060 ---> ACK sip:2249@10.0.2.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK2107005634 From: "Unidata" ;tag=327081027 To: ;tag=as69233e59 Call-ID: 2138401834@10.251.18.235 CSeq: 21 ACK Max-Forwards: 70 User-Agent: WPU-7700-v2.8.0/0 Content-Length: 0 <-------------> [Apr 28 13:39:27] --- (9 headers 0 lines) --- [Apr 28 13:39:30] <--- SIP read from UDP:10.0.2.200:5060 ---> INVITE sip:89019@10.0.2.22:5060 SIP/2.0 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 1 INVITE Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f56-391e6a48-2529f998 Max-Forwards: 70 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 P-Asserted-Identity: "Baldus Stefan" Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 235 v=0 o=- 30355 2 IN IP4 10.0.2.200 s=- t=0 0 m=audio 5208 RTP/AVP 8 0 18 101 111 c=IN IP4 10.0.2.133 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=sendrecv <-------------> [Apr 28 13:39:30] --- (15 headers 12 lines) --- [Apr 28 13:39:30] Sending to 10.0.2.200:5060 (no NAT) [Apr 28 13:39:30] Found RTP audio format 8 [Apr 28 13:39:30] Found RTP audio format 0 [Apr 28 13:39:30] Found RTP audio format 18 [Apr 28 13:39:30] Found RTP audio format 101 [Apr 28 13:39:30] Found RTP audio format 111 [Apr 28 13:39:30] Found audio description format telephone-event for ID 101 [Apr 28 13:39:30] Found audio description format X-nt-inforeq for ID 111 [Apr 28 13:39:30] Capabilities: us - 0x8 (alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Apr 28 13:39:30] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [Apr 28 13:39:30] Peer audio RTP is at port 10.0.2.133:5208 [Apr 28 13:39:30] <--- Transmitting (no NAT) to 10.0.2.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f56-391e6a48-2529f998;received=10.0.2.200 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> [Apr 28 13:39:30] Audio is at 5060 [Apr 28 13:39:30] Adding codec 0x8 (alaw) to SDP [Apr 28 13:39:30] <--- Reliably Transmitting (no NAT) to 10.0.2.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f56-391e6a48-2529f998;received=10.0.2.200 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 197 v=0 o=root 55210502 55210503 IN IP4 10.0.2.22 s=Asterisk PBX 1.8.3.2 c=IN IP4 10.0.2.22 t=0 0 m=audio 13628 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 28 13:39:30] <--- SIP read from UDP:10.0.2.200:5060 ---> ACK sip:89019@10.0.2.22:5060 SIP/2.0 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f56-391e6a7a-615ed50a Max-Forwards: 70 User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> [Apr 28 13:39:33] --- (10 headers 0 lines) --- [Apr 28 13:39:33] Really destroying SIP dialog '1a5f206c3a5c786212b8b7470bcc4a93@sipgw1.' Method: OPTIONS [Apr 28 13:39:34] <--- SIP read from UDP:10.0.2.200:5060 ---> INVITE sip:89019@10.0.2.22:5060 SIP/2.0 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 2 INVITE Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f5a-391e788a-45d4d183 Max-Forwards: 70 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 P-Asserted-Identity: "Baldus Stefan" Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 536 v=0 o=- 30355 3 IN IP4 10.0.2.200 s=- t=0 0 m=audio 5222 RTP/AVP 8 0 18 101 111 c=IN IP4 10.250.141.21 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=sendrecv m=audio 5222 RTP/SAVP 8 0 18 101 111 c=IN IP4 10.250.141.21 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5VT9czSiPF5v/8xwGNkHFx8STxjyIePzThdZAHwG|2^031|001977971830:004 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=sendrecv <-------------> [Apr 28 13:39:34] --- (15 headers 21 lines) --- [Apr 28 13:39:34] Sending to 10.0.2.200:5060 (no NAT) [Apr 28 13:39:34] Found RTP audio format 8 [Apr 28 13:39:34] Found RTP audio format 0 [Apr 28 13:39:34] Found RTP audio format 18 [Apr 28 13:39:34] Found RTP audio format 101 [Apr 28 13:39:34] Found RTP audio format 111 [Apr 28 13:39:34] Found audio description format telephone-event for ID 101 [Apr 28 13:39:34] Found audio description format X-nt-inforeq for ID 111 [Apr 28 13:39:34] Found RTP audio format 8 [Apr 28 13:39:34] Found RTP audio format 0 [Apr 28 13:39:34] Found RTP audio format 18 [Apr 28 13:39:34] Found RTP audio format 101 [Apr 28 13:39:34] Found RTP audio format 111 [Apr 28 13:39:34] ERROR[2988]: chan_sip.c:27991 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Apr 28 13:39:34] Found audio description format telephone-event for ID 101 [Apr 28 13:39:34] Found audio description format X-nt-inforeq for ID 111 [Apr 28 13:39:34] WARNING[2988]: chan_sip.c:8412 process_sdp: Can't provide secure audio requested in SDP offer [Apr 28 13:39:34] <--- Reliably Transmitting (no NAT) to 10.0.2.200:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f5a-391e788a-45d4d183;received=10.0.2.200 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> [Apr 28 13:39:34] <--- SIP read from UDP:10.0.2.200:5060 ---> ACK sip:89019@10.0.2.22:5060 SIP/2.0 From: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 To: "Unidata";tag=as462aa110 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 2 ACK Via: SIP/2.0/UDP 10.0.2.200:5060;branch=z9hG4bK-e9f5a-391e788a-45d4d183 Max-Forwards: 70 User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> [Apr 28 13:39:34] --- (11 headers 0 lines) --- [Apr 28 13:39:41] <--- SIP read from UDP:10.251.18.235:5060 ---> BYE sip:2249@10.0.2.22:5060 SIP/2.0 Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK1053285485 From: "Unidata" ;tag=327081027 To: ;tag=as69233e59 Call-ID: 2138401834@10.251.18.235 CSeq: 22 BYE Max-Forwards: 70 User-Agent: WPU-7700-v2.8.0/0 Content-Length: 0 <-------------> [Apr 28 13:39:41] --- (9 headers 0 lines) --- [Apr 28 13:39:41] Sending to 10.251.18.235:5060 (no NAT) [Apr 28 13:39:41] Scheduling destruction of SIP dialog '2138401834@10.251.18.235' in 40576 ms (Method: BYE) [Apr 28 13:39:41] <--- Transmitting (no NAT) to 10.251.18.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.251.18.235:5060;branch=z9hG4bK1053285485;received=10.251.18.235 From: "Unidata" ;tag=327081027 To: ;tag=as69233e59 Call-ID: 2138401834@10.251.18.235 CSeq: 22 BYE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> [Apr 28 13:39:41] Scheduling destruction of SIP dialog '5349ccaa7501889a32a6f8a37cd5185a@sipgw1.' in 6400 ms (Method: ACK) [Apr 28 13:39:41] set_destination: Parsing for address/port to send to [Apr 28 13:39:41] Reliably Transmitting (no NAT) to 10.0.2.200:5060: BYE sip:2249;phone-context=UnknownUnknown@vs10.sip:5060;maddr=10.0.2.200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK25824029 Max-Forwards: 70 From: "Unidata";tag=as462aa110 To: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 103 BYE User-Agent: Asterisk PBX 1.8.3.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Apr 28 13:39:41] == Spawn extension (dialout-nopin, 2249, 1) exited non-zero on 'SIP/89019-00000000' [Apr 28 13:39:41] <--- SIP read from UDP:10.0.2.200:5060 ---> SIP/2.0 200 OK From: "Unidata";tag=as462aa110 To: ;tag=1a30e730-c802000a-13c4-40030-e9f50-2b2b1d6d-e9f50 Call-ID: 5349ccaa7501889a32a6f8a37cd5185a@sipgw1. CSeq: 103 BYE Via: SIP/2.0/UDP 10.0.2.22:5060;branch=z9hG4bK25824029 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Content-Length: 0 <-------------> [Apr 28 13:39:41] --- (9 headers 0 lines) --- [Apr 28 13:39:41] SIP Response message for INCOMING dialog BYE arrived [Apr 28 13:39:41] Really destroying SIP dialog '5349ccaa7501889a32a6f8a37cd5185a@sipgw1.' Method: ACK test-ast*CLI> test-ast*CLI>