asterisk1*CLI> <-- SIP read from 208.239.76.170:5060: INVITE sip:s@38.113.114.224;useradd=38.113.114.224;userport=5060 SIP/2.0 Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu100051njgoc2g0.1 From: "unknown" ;tag=SDokoo601-as058ae75d To: Contact: Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 CSeq: 102 INVITE User-Agent: IPX Softswitch 2.0 Max-Forwards: 69 Remote-Party-ID: "unknown" ;privacy=off;screen=no Date: Wed, 29 Aug 2007 01:33:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-UUID: 1188351203.899855.5302 Content-Type: application/sdp Content-Length: 556 v=0 o=root 7130 7130 IN IP4 208.239.76.170 s=session c=IN IP4 208.239.76.170 t=0 0 m=audio 56864 RTP/AVP 3 4 18 111 97 0 8 10 101 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 63518 RTP/AVP 26 103 34 31 a=rtpmap:26 JPEG/90000 a=rtpmap:103 h263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 --- (15 headers 23 lines) --- Using INVITE request as basis request - SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 Sending to 208.239.76.170 : 5060 (non-NAT) Found peer '8182323408_OUT' Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 111 Found RTP audio format 97 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 10 Found RTP audio format 101 Peer audio RTP is at port 208.239.76.170:56864 Found description format GSM Found description format G723 Found description format G729 Found description format G726-32 Found description format iLBC Found description format PCMU Found description format PCMA Found description format L16 Found description format telephone-event Found description format JPEG Found description format h263-1998 Found description format H263 Found description format H261 Capabilities: us - 0x107 (g723|gsm|ulaw|g729), peer - audio=0x55f (g723|gsm|ulaw|alaw|g726|slin|g729|ilbc)/video=0x0 (nothing), combined - 0x107 (g723|gsm|ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in from-pstn (domain 38.113.114.224;useradd=38.113.114.224;userport=5060) list_route: hop: Transmitting (no NAT) to 208.239.76.170:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu100051njgoc2g0.1;received=208.239.76.170 From: "unknown" ;tag=SDokoo601-as058ae75d To: Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing NoOp("SIP/7665699-b79043c0", "No DID or CID Match") in new stack -- Executing Answer("SIP/7665699-b79043c0", "") in new stack We're at 38.113.114.224 port 12220 Adding codec 0x4 (ulaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 208.239.76.170:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu100051njgoc2g0.1;received=208.239.76.170 From: "unknown" ;tag=SDokoo601-as058ae75d To: ;tag=as10af7068 Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 312 v=0 o=root 6417 6417 IN IP4 38.113.114.224 s=session c=IN IP4 38.113.114.224 t=0 0 m=audio 12220 RTP/AVP 0 4 18 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/7665699-b79043c0", "2") in new stack asterisk1*CLI> <-- SIP read from 208.239.76.170:5060: ACK sip:s@38.113.114.224 SIP/2.0 Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu1010shhd0bv7g0.1 From: "unknown" ;tag=SDokoo601-as058ae75d To: ;tag=as10af7068 Contact: Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 CSeq: 102 ACK User-Agent: IPX Softswitch 2.0 Max-Forwards: 69 Remote-Party-ID: "unknown" ;privacy=off;screen=no Content-Length: 0 --- (11 headers 0 lines) --- -- Executing Playback("SIP/7665699-b79043c0", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing SayAlpha("SIP/7665699-b79043c0", "") in new stack 12 headers, 0 lines Reliably Transmitting (NAT) to 76.175.234.137:5060: OPTIONS sip:9000@192.168.1.137:5060;rinstance=1fcecec2909e2ce2 SIP/2.0 Via: SIP/2.0/UDP 38.113.114.224:5060;branch=z9hG4bK6edc8add;rport From: "Unknown" ;tag=as5759b581 To: Contact: Call-ID: 332d6c18319b90b04320551732c5fab7@38.113.114.224 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Aug 2007 01:33:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 76.175.234.137:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.113.114.224:5060;branch=z9hG4bK6edc8add;rport=5060 Contact: To: ;tag=b270d37b From: "Unknown";tag=as5759b581 Call-ID: 332d6c18319b90b04320551732c5fab7@38.113.114.224 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS User-Agent: Idefisk Allow-Events: message-summary Content-Length: 0 --- (13 headers 0 lines) --- Destroying call '332d6c18319b90b04320551732c5fab7@38.113.114.224' Scheduling destruction of call 'SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 208.239.76.170, port 5060 Reliably Transmitting (no NAT) to 208.239.76.170:5060: BYE sip:3104991266@208.239.76.170:5060 SIP/2.0 Via: SIP/2.0/UDP 38.113.114.224:5060;branch=z9hG4bK7148ffb2;rport From: ;tag=as10af7068 To: "unknown" ;tag=SDokoo601-as058ae75d Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 208.239.76.170:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.113.114.224:5060;received=38.113.114.224;branch=z9hG4bK7148ffb2;rport=5060 From: ;tag=as10af7068 To: "unknown" ;tag=SDokoo601-as058ae75d Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2 CSeq: 102 BYE User-Agent: IPX Softswitch 2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- (11 headers 0 lines) --- Destroying call 'SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2' asterisk1*CLI> <-- SIP read from 76.175.234.137:5060: --- (0 headers 1 lines) ---