============ To: Contact: Call-ID: 4fd908680d46bcd35ff3901c51bc0120@82.83.56.91 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 30 Jul 2004 20:40:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 314 v=0 o=root 32131 32131 IN IP4 82.83.56.91 s=session c=IN IP4 82.83.56.91 t=0 0 m=audio 15080 RTP/AVP 8 0 2 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 217.10.79.9:5060 -- Called 07141220856@sipgate Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Peer RTP is at port 217.10.64.78:0 Found description format PCMA Found description format PCMU Found description format GSM Found description format iLBC Found description format G726-32 Found description format telephone-event Capabilities: us - 0x41e(GSM|ULAW|ALAW|G726|ILBC), peer - audio=0x41e(GSM|ULAW|ALAW|G726|ILBC)/video=0x0(EMPTY), combined - 0x41e(GSM|ULAW|ALAW|G726|ILBC) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) -- SIP/sipgate-76e5 is making progress passing it to SIP/2112-495b Reliably Transmitting: CANCEL sip:497141220856@217.10.64.78 SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b9e12cc From: "Florian" ;tag=as1c3c2263 To: Contact: Call-ID: 4fd908680d46bcd35ff3901c51bc0120@82.83.56.91 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.10.79.9:5060 Scheduling destruction of call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' in 15000 ms == Spawn extension (out, 907141220856, 3) exited non-zero on 'SIP/2112-495b' Transmitting: ACK sip:497141220856@217.10.64.78 SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b9e12cc From: "Florian" ;tag=as1c3c2263 To: ;tag=cbf5cb1d0d4e31526039b4f3671ccf51-3d18 Contact: Call-ID: 4fd908680d46bcd35ff3901c51bc0120@82.83.56.91 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.10.79.9:5060 Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:217.10.79.9 SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK2eefa4de From: "asterisk" ;tag=as7c4322e5 To: Contact: Call-ID: 6d3771aa1377fbf249c866b5767becfe@82.83.56.91 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Fri, 30 Jul 2004 20:41:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 217.10.79.9:5060 Destroying call '6d3771aa1377fbf249c866b5767becfe@82.83.56.91' 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK388496a5 From: ;tag=as7cc829e2 To: Call-ID: 6ed4586d1befa53f5f12e16a6d22e667@192.168.99.11 CSeq: 108 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.10.79.9:5060 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b70e7c4 From: ;tag=as7cc829e2 To: Call-ID: 6ed4586d1befa53f5f12e16a6d22e667@192.168.99.11 CSeq: 109 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="8888888", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="410ab3a95b427f0b044db27cdada37d268edcfee", response="f4b21e167ad3b177beddceac6aea0567", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 (no NAT) to 217.10.79.9:5060 Destroying call '6ed4586d1befa53f5f12e16a6d22e667@192.168.99.11' Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91' Destroying call '4fd908680d46bcd35ff3901c51bc0120@82.83.56.91'