Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 64.2.142.17:5060: INVITE sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2fa31702;rport From: "102" ;tag=as55937ff3 To: Contact: Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "102" ;privacy=off;screen=no Date: Sat, 29 Dec 2007 23:31:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 473 v=0 o=root 2926 2926 IN IP4 yyy.yyy.yyy.yyy s=session c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 11844 RTP/AVP 0 3 8 112 5 10 7 97 111 101 -- Re-regist ration for myusername@inbound5.vitelity.net [Dec 29 18:32:56] NOTICE[2933]: chan_sip.c:12113 handle_response_register: Outb ound Registration: Expiry for inbound5.vitelity.net is 60 sec (Scheduling reregi stration in 45 s)-G726-32/8000 Really destroying SIP dialog '567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy' M ethod: INVITE16/8000 litclinux*CLI> a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- litclinux*CLI> <--- SIP read from 64.2.142.17:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2fa31702;received=yyy.yyy.yyy.yyy;rport=5060 From: "102" ;tag=as55937ff3 To: ;tag=as4abecb20 Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a8efe03" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 64.2.142.17:5060: ACK sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2fa31702;rport From: "102" ;tag=as55937ff3 To: ;tag=as4abecb20 Contact: Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "102" ;privacy=off;screen=no Content-Length: 0 --- Audio is at yyy.yyy.yyy.yyy port 11844 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 64.2.142.17:5060: INVITE sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;rport From: "102" ;tag=as55937ff3 To: Contact: Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "102" ;privacy=off;screen=no Proxy-Authorization: Digest username="myusername", realm="asterisk", algorithm=MD5, uri="sip:1xxxxxxxxxx@outbound1.vitelity.net", nonce="0a8efe03", response="82d12 466c5c546b18f03534b55feb6ef", opaque="" Date: Sat, 29 Dec 2007 23:31:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 473 v=0 o=root 2926 2927 IN IP4 yyy.yyy.yyy.yyy s=session c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 11844 RTP/AVP 0 3 8 112 5 10 7 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- litclinux*CLI> <--- SIP read from 64.2.142.17:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060 From: "102" ;tag=as55937ff3 To: Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Dec 29 18:32:11] NOTICE[2933]: chan_sip.c:7145 sip_reregister: -- Re-regist ration for myusername@inbound5.vitelity.net [Dec 29 18:32:11] NOTICE[2933]: chan_sip.c:12113 handle_response_register: Outb ound Registration: Expiry for inbound5.vitelity.net is 60 sec (Scheduling reregi stration in 45 s) Reliably Transmitting (NAT) to 64.2.142.17:5060: OPTIONS sip:outbound1.vitelity.net SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK0aa2be62;rport From: "asterisk" ;tag=as5032586a To: Contact: Call-ID: 4aa917382181b8ff6e1c90d7275db8bd@yyy.yyy.yyy.yyy CSeq: 102 OPTIONS [Dec 29 18:32:56] NOTICE[2933]: chan_sip.c:7145 sip_reregister: Max-Forwards: 70 Date: Sat, 29 Dec 2007 23:32:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- litclinux*CLI> <--- SIP read from 64.2.142.17:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK0aa2be62;received=yyy.yyy.yyy.yyy;rport=5060 From: "asterisk" ;tag=as5032586a To: ;tag=as09be9d34 Call-ID: 4aa917382181b8ff6e1c90d7275db8bd@yyy.yyy.yyy.yyy CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '4aa917382181b8ff6e1c90d7275db8bd@yyy.yyy.yyy.yyy' Method: OPTIONS litclinux*CLI> <--- SIP read from 64.2.142.17:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060 From: "102" ;tag=as55937ff3 To: ;tag=as2b448282 Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 5396 5396 IN IP4 64.2.142.17 s=session c=IN IP4 64.2.142.17 t=0 0 m=audio 12534 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (11 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 64.2.142.17:12534 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format telephone-event for ID 101 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|spee x|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw| alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephon e-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 64.2.142.17:12534 Scheduling destruction of SIP dialog '567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 64.2.142.17:5060: CANCEL sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;rport From: "102" ;tag=as55937ff3 To: Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 emote-Party-ID: "102" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy' in 6400 ms (Method: INVITE) litclinux*CLI> <--- SIP read from 64.2.142.17:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060 From: "102" ;tag=as55937ff3 To: ;tag=as2b448282 Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 X-Asterisk-HangupCause: Network out of order <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 64.2.142.17:5060: ACK sip:1xxxxxxxxxx@outbound1.vitelity.net SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;rport From: "102" ;tag=as55937ff3 To: ;tag=as2b448282 Contact: Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "102" ;privacy=off;screen=no Content-Length: 0 --- litclinux*CLI> <--- SIP read from 64.2.142.17:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5c65a47d;received=yyy.yyy.yyy.yyy;rport=5060 From: "102" ;tag=as55937ff3 To: ;tag=as2b448282 Call-ID: 567f8a5d6736722b0b366a5338e6c770@yyy.yyy.yyy.yyy CSeq: 103 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Network out of order <-------------> --- (11 headers 0 lines) --- litclinux*CLI>