aries*CLI> <-- SIP read from 66.250.69.16:5060: INVITE sip:8006170141@host41.com SIP/2.0 Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK1c028e1f From: "4072930335" ;tag=as6b37a291 To: Contact: Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16 CSeq: 102 INVITE User-Agent: asterlink/1.2 Date: Tue, 11 Jul 2006 17:29:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 469 v=0 o=root 9978 9978 IN IP4 66.250.69.16 s=session c=IN IP4 66.250.69.16 t=0 0 m=audio 23940 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (12 headers 20 lines)--- Using INVITE request as basis request - 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16 Sending to 66.250.69.16 : 5060 (non-NAT) Found peer 'asterlink-switch-06' Found RTP audio format 10 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 5 Found RTP audio format 7 Found RTP audio format 18 Found RTP audio format 110 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 66.250.69.16:23940 Found description format L16 Found description format G723 Found description format GSM Found description format PCMU Found description format PCMA Found description format G726-32 Found description format DVI4 Found description format LPC Found description format G729 Found description format speex Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 8006170141 in incoming (domain host41.com) list_route: hop: Transmitting (no NAT) to 66.250.69.16:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK1c028e1f;received=66.250.69.16 From: "4072930335" ;tag=as6b37a291 To: Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Destroying call '29e216df2bea67bf73615d3210d9773d@72.232.182.13' Destroying call '63a8663921d8836d4e4fd65a636b6526@72.232.182.13' We're at 72.232.182.13 port 16618 Adding codec 0x2 (gsm) to SDP 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 66.250.69.16:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK1c028e1f;received=66.250.69.16 From: "4072930335" ;tag=as6b37a291 To: ;tag=as522fefca Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 3534 3534 IN IP4 72.232.182.13 s=session c=IN IP4 72.232.182.13 t=0 0 m=audio 16618 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- aries*CLI> <-- SIP read from 66.250.69.16:5060: ACK sip:8006170141@72.232.182.13 SIP/2.0 Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK0de25ef0 From: "4072930335" ;tag=as6b37a291 To: ;tag=as522fefca Contact: Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16 CSeq: 102 ACK User-Agent: asterlink/1.2 Content-Length: 0 --- (9 headers 0 lines)--- We're at 72.232.182.13 port 14066 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 72.40.10.150:5060: INVITE sip:200@72.40.10.150:5060 SIP/2.0 Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK4fd8c5e7;rport From: "4072930335" ;tag=as4aa16d7d To: Contact: Call-ID: 08c965242994ae18514f92cd2c964b17@72.232.182.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 11 Jul 2006 17:30:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 3534 3534 IN IP4 72.232.182.13 s=session c=IN IP4 72.232.182.13 t=0 0 m=audio 14066 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 11 13:30:46 WARNING[1949]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 aries*CLI> <-- SIP read from 72.40.10.150:5060: SIP/2.0 404 Not Found To: ;tag=656eae54128438bbi0 From: "4072930335" ;tag=as4aa16d7d Call-ID: 08c965242994ae18514f92cd2c964b17@72.232.182.13 CSeq: 102 INVITE Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK4fd8c5e7 Server: Linksys/PAP2-3.1.3(LS) Content-Length: 0 --- (8 headers 0 lines)--- Transmitting (no NAT) to 72.40.10.150:5060: ACK sip:200@72.40.10.150:5060 SIP/2.0 Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK4fd8c5e7;rport From: "4072930335" ;tag=as4aa16d7d To: ;tag=656eae54128438bbi0 Contact: Call-ID: 08c965242994ae18514f92cd2c964b17@72.232.182.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '08c965242994ae18514f92cd2c964b17@72.232.182.13' aries*CLI>