Sip read: INVITE sip:08450043965@321.321.321.321 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK0a55.8187a175.0 Via: SIP/2.0/UDP 123.123.123.123:5060 From: "anonymous" ;tag=8729A358-45A To: Date: Tue, 22 Feb 2005 15:27:06 gmt Call-ID: B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: ;party=calling;screen=yes;privacy=full Timestamp: 1109086026 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 348 v=0 o=CiscoSystemsSIP-GW-UserAgent 9120 9119 IN IP4 123.123.123.123 s=SIP Call c=IN IP4 123.123.123.123 t=0 0 m=audio 19134 RTP/AVP 4 18 3 2 8 0 c=IN IP4 123.123.123.123 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 19 headers, 15 lines Using latest request as basis request Sending to 123.123.123.123 : 5060 (non-NAT) Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 123.123.123.123:19134 Found description format G723 Found description format G729 Found description format GSM Found description format G726-32 Found description format PCMA Found description format PCMU Capabilities: us - 0x40a (gsm|alaw|ilbc), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Found peer 'magrathea-in' Looking for 08450043965 in incoming_numbers list_route: hop: list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK0a55.8187a175.0 Via: SIP/2.0/UDP 123.123.123.123:5060 From: "anonymous" ;tag=8729A358-45A To: ;tag=as7a8b012e Call-ID: B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 123.123.123.123:5060 -- Executing Wait("SIP/123.123.123.123-4093a758", "4") in new stack voip-1*CLI> Sip read: 0 headers, 0 lines voip-1*CLI> Sip read: 0 headers, 0 lines -- Executing Answer("SIP/123.123.123.123-4093a758", "") in new stack We're at 321.321.321.321 port 16836 Answering with preferred capability 0x8 (alaw) Answering with capability 0x100 (g729) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK0a55.8187a175.0 Via: SIP/2.0/UDP 123.123.123.123:5060 Record-Route: From: "anonymous" ;tag=8729A358-45A To: ;tag=as7a8b012e Call-ID: B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 192 v=0 o=root 13400 13400 IN IP4 321.321.321.321 s=session c=IN IP4 321.321.321.321 t=0 0 m=audio 16836 RTP/AVP 8 18 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - to 123.123.123.123:5060 -- Executing DigitTimeout("SIP/123.123.123.123-4093a758", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("SIP/123.123.123.123-4093a758", "10") in new stack -- Set Response Timeout to 10 -- Executing BackGround("SIP/123.123.123.123-4093a758", "/tmp/asterisk-recording") in new stack -- Playing '/tmp/asterisk-recording' (language 'en') voip-1*CLI> Sip read: ACK sip:08450043965@321.321.321.321:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 123.123.123.123;branch=0 Via: SIP/2.0/UDP 123.123.123.123:5060 From: "anonymous" ;tag=8729A358-45A To: ;tag=as7a8b012e Date: Tue, 22 Feb 2005 15:27:06 gmt Call-ID: B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123 Max-Forwards: 5 Content-Length: 0 CSeq: 101 ACK 11 headers, 0 lines Sip read: 0 headers, 0 lines == CDR updated on SIP/123.123.123.123-4093a758 -- Executing SetGroup("SIP/123.123.123.123-4093a758", "asyouneed") in new stack -- Executing CheckGroup("SIP/123.123.123.123-4093a758", "2") in new stack -- Executing DBput("SIP/123.123.123.123-4093a758", "asyouneed/1=123123123123") in new stack -- DBput: family=asyouneed, key=1, value=123123123123 -- Executing Dial("SIP/123.123.123.123-4093a758", "SIP/user-1&SIP/user-4|20|tr") in new stack We're at 321.321.321.321 port 18958 Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:user-1@192.168.0.7 SIP/2.0 Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: Contact: Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 22 Feb 2005 15:27:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 224 v=0 o=root 13400 13400 IN IP4 321.321.321.321 s=session c=IN IP4 321.321.321.321 t=0 0 m=audio 18958 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 82.45.243.58:1026 -- Called user-1 Feb 22 15:27:14 WARNING[13400]: chan_sip.c:1404 create_addr: No such host: user-4 Destroying call '3ab17cc100bd600f27a3a9956d669a22@321.321.321.321' Feb 22 15:27:14 NOTICE[13400]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) voip-1*CLI> Sip read: 0 headers, 0 lines voip-1*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 INVITE User-Agent: Grandstream HT487 1.0.5.21 Content-Length: 0 8 headers, 0 lines voip-1*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: ;tag=b6ef757a0392f401 Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 INVITE User-Agent: Grandstream HT487 1.0.5.21 Content-Length: 0 8 headers, 0 lines -- SIP/user-1-e691 is ringing voip-1*CLI> Sip read: BYE sip:08450043965@321.321.321.321:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bKd955.355327e2.0 Via: SIP/2.0/UDP 123.123.123.123:5060 From: "anonymous" ;tag=8729A358-45A To: ;tag=as7a8b012e Date: Tue, 22 Feb 2005 15:27:06 gmt Call-ID: B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 5 Timestamp: 1109086036 CSeq: 102 BYE Content-Length: 0 13 headers, 0 lines Sending to 123.123.123.123 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bKd955.355327e2.0 Via: SIP/2.0/UDP 123.123.123.123:5060 Record-Route: From: "anonymous" ;tag=8729A358-45A To: ;tag=as7a8b012e Call-ID: B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 123.123.123.123:5060 Reliably Transmitting: CANCEL sip:user-1@192.168.0.7 SIP/2.0 Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: Contact: Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 82.45.243.58:1026 Scheduling destruction of call '690992da18de18440d343ee96286647f@321.321.321.321' in 15000 ms == Spawn extension (incoming_numbers, 1, 4) exited non-zero on 'SIP/123.123.123.123-4093a758' voip-1*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: ;tag=22c043bea9bbce19 Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 CANCEL User-Agent: Grandstream HT487 1.0.5.21 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 11 headers, 0 lines Destroying call 'B25A938-841D11D9-85B8B54A-A9B6819A@123.123.123.123' voip-1*CLI> Sip read: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: ;tag=b5b39ae72802efdf Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 INVITE User-Agent: Grandstream HT487 1.0.5.21 Content-Length: 0 8 headers, 0 lines Transmitting: ACK sip:user-1@192.168.0.7 SIP/2.0 Via: SIP/2.0/UDP 321.321.321.321:5060;branch=z9hG4bK33587eec;rport From: "anonymous" ;tag=as285397aa To: ;tag=b5b39ae72802efdf Contact: Call-ID: 690992da18de18440d343ee96286647f@321.321.321.321 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 82.45.243.58:1026 Destroying call '690992da18de18440d343ee96286647f@321.321.321.321'