<--- SIP read from UDP:10.4.24.11:33737 ---> INVITE sip:8000;phone-context=cdp.udp@contoso.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK3a1545f15fd3cd84ff515f5f.1 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a2-da2254a0-3073a4e0;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 1 INVITE Contact: Max-forwards: 69 Supported: 100rel,x-nortel-sipvc,replaces,timer User-agent: Nortel CS1000 SIP GW release_6.0 version_ssLinux-6.00.18 P-asserted-identity: Privacy: none History-info: ;index=1 X-nt-corr-id: 000000f00b120d030c@00c08b0ad2ec-e4f35711 Min-se: 0 Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: multipart/mixed ;boundary=unique-boundary-1 Content-Length: 754 --unique-boundary-1 Content-Type: application/sdp v=0 o=- 310 1 IN IP4 10.4.24.10 s=- t=0 0 m=audio 5200 RTP/AVP 0 8 101 111 c=IN IP4 10.4.24.95 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv --unique-boundary-1 Content-Type: application/x-nt-mcdn-frag-hex;version=ssLinux-6.00.18;base=x2611 Content-Disposition: signal;handling=optional 05009801 0107130081900000a200 09090f00e9a0830001001800 1315070011fa0f00a10d02010102020100cc0400007e1d00 1e0403008183 --unique-boundary-1 Content-Type: application/x-nt-epid-frag-hex;version=ssLinux-6.00.18;base=x2611 Content-Disposition: signal;handling=optional 011201 00:24:e8:79:45:0a --unique-boundary-1-- <-------------> --- (20 headers 32 lines) --- == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 Sending to 10.4.24.11:5060 (no NAT) Using INVITE request as basis request - 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 No matching peer for '6677;phone-context=cdp.udp' from '10.4.24.11:33737' == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 111 Found audio description format telephone-event for ID 101 Found audio description format X-nt-inforeq for ID 111 Capabilities: us - 0x38160e (gsm|ulaw|alaw|speex|ilbc|g722|h263|h263p|h264), peer - audio=0x80c (ulaw|alaw|g726)/video=0x0 (nothing)/text=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 10.4.24.95:5200 Peer doesn't provide video Looking for 8000;phone-context=cdp.udp in from-sip-external (domain contoso.com) list_route: hop: <--- Transmitting (no NAT) to 10.4.24.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK3a1545f15fd3cd84ff515f5f.1;received=10.4.24.11 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a2-da2254a0-3073a4e0;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 1 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [8000;phone-context=cdp.udp@from-sip-external:1] NoOp("SIP/contoso.com-00000016", "Received incoming SIP connection from unknown peer to 8000;phone-context=cdp.udp") in new stack -- Executing [8000;phone-context=cdp.udp@from-sip-external:2] Set("SIP/contoso.com-00000016", "DID=8000;phone-context=cdp.udp") in new stack -- Executing [8000;phone-context=cdp.udp@from-sip-external:3] Goto("SIP/contoso.com-00000016", "s,1") in new stack -- Goto (from-sip-external,s,1) -- Executing [s@from-sip-external:1] GotoIf("SIP/contoso.com-00000016", "1?checklang:noanonymous") in new stack -- Goto (from-sip-external,s,2) -- Executing [s@from-sip-external:2] GotoIf("SIP/contoso.com-00000016", "0?setlanguage:from-trunk,8000;phone-context=cdp.udp,1") in new stack -- Goto (from-trunk,8000;phone-context=cdp.udp,1) -- Executing [8000;phone-context=cdp.udp@from-trunk:1] NoOp("SIP/contoso.com-00000016", "Catch-All DID Match - Found 8000;phone-context=cdp.udp - You probably want a DID for this.") in new stack -- Executing [8000;phone-context=cdp.udp@from-trunk:2] Goto("SIP/contoso.com-00000016", "ext-did,s,1") in new stack -- Goto (ext-did,s,1) -- Executing [s@ext-did:1] Set("SIP/contoso.com-00000016", "__FROM_DID=s") in new stack -- Executing [s@ext-did:2] ExecIf("SIP/contoso.com-00000016", "1 ?Set(CALLERID(name)=6677)") in new stack -- Executing [s@ext-did:3] Set("SIP/contoso.com-00000016", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [s@ext-did:4] Set("SIP/contoso.com-00000016", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [s@ext-did:5] Goto("SIP/contoso.com-00000016", "custom-meetme3,s,1") in new stack -- Goto (custom-meetme3,s,1) -- Executing [s@custom-meetme3:1] Answer("SIP/contoso.com-00000016", "") in new stack Audio is at 5060 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 10.4.24.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK3a1545f15fd3cd84ff515f5f.1;received=10.4.24.11 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a2-da2254a0-3073a4e0;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: ;tag=as0ac3ddfc Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 1 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 254 v=0 o=root 1679432374 1679432374 IN IP4 10.4.16.45 s=Asterisk PBX 1.8.0 c=IN IP4 10.4.16.45 t=0 0 m=audio 10238 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:10.4.24.11:33737 ---> ACK sip:8000;phone-context=cdp.udp@10.4.16.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK86666549e8b71cf2fce8bfd9.1 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a2-da2254a9-639cb2b2;received=10.4.24.10 From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: ;tag=as0ac3ddfc Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 1 ACK Contact: Max-forwards: 70 User-agent: Nortel CS1000 SIP GW release_6.0 version_ssLinux-6.00.18 X-nt-corr-id: 000000f00b120d030c@00c08b0ad2ec-e4f35711 Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:10.4.24.11:33737 ---> OPTIONS sip:8000;phone-context=cdp.udp@10.4.16.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK24dca24d8a5e555327bcf562.1 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a2-da2254aa-5f7338f8;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: ;tag=as0ac3ddfc Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 2 OPTIONS Contact: Max-forwards: 69 Supported: 100rel,x-nortel-sipvc,replaces,timer User-agent: Nortel CS1000 SIP GW release_6.0 version_ssLinux-6.00.18 X-nt-corr-id: 000000f00b120d030c@00c08b0ad2ec-e4f35711 Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Length: 0 <-------------> --- (15 headers 0 lines) --- <--- Transmitting (no NAT) to 10.4.24.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK24dca24d8a5e555327bcf562.1;received=10.4.24.11 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a2-da2254aa-5f7338f8;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: ;tag=as0ac3ddfc Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 2 OPTIONS Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <------------> -- Executing [s@custom-meetme3:2] Wait("SIP/contoso.com-00000016", "1") in new stack -- Executing [s@custom-meetme3:3] MeetMe("SIP/contoso.com-00000016", "") in new stack -- Playing 'conf-getconfno.ulaw' (language 'en') <--- SIP read from UDP:10.4.24.11:33737 ---> BYE sip:8000;phone-context=cdp.udp@10.4.16.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK19b9bd40dc465d3777047ddb.1 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a5-da225efa-3ef2dd9e;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: ;tag=as0ac3ddfc Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 3 BYE Max-forwards: 69 Supported: 100rel,x-nortel-sipvc,replaces,timer User-agent: Nortel CS1000 SIP GW release_6.0 version_ssLinux-6.00.18 X-nt-corr-id: 000000f00b120d030c@00c08b0ad2ec-e4f35711 Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length: 224 v=0 o=- 310 1 IN IP4 10.4.24.10 s=- t=0 0 m=audio 5200 RTP/AVP 0 8 101 111 c=IN IP4 10.4.24.95 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv <-------------> --- (15 headers 12 lines) --- Sending to 10.4.24.11:5060 (no NAT) Scheduling destruction of SIP dialog '8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.4.24.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.24.11:5060;branch=z9hG4bK19b9bd40dc465d3777047ddb.1;received=10.4.24.11 Via: SIP/2.0/TCP 10.4.24.10:5060;branch=z9hG4bK-37d7a5-da225efa-3ef2dd9e;received=10.4.24.10 Record-Route: From: ;tag=7f25aa8-a18040a-13c4-40030-37d7a2-21727553-37d7a2 To: ;tag=as0ac3ddfc Call-ID: 8ef2998-a18040a-13c4-40030-37d7a2-62374f5d-37d7a2 CSeq: 3 BYE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (custom-meetme3, s, 3) exited non-zero on 'SIP/contoso.com-00000016' -- Executing [h@custom-meetme3:1] ExecIf("SIP/contoso.com-00000016", "0?NoOp:NoCDR()") in new stack -- Executing [h@custom-meetme3:2] Set("SIP/contoso.com-00000016", "CDR(bookId)=") in new stack -- Executing [h@custom-meetme3:3] Set("SIP/contoso.com-00000016", "CDR(CIDnum)=6677") in new stack -- Executing [h@custom-meetme3:4] Set("SIP/contoso.com-00000016", "CDR(CIDname)=6677") in new stack Really destroying SIP dialog '8ef27d0-a18040a-13c4-40030-37d782-4fc4931c-37d782' Method: BYE