Connected to Asterisk 1.2.9.1 currently running on aries (pid = 29933) aries*CLI> sip debug SIP Debugging enabled aries*CLI> <-- SIP read from 66.250.69.13:5060: INVITE sip:8006170141@host41.com SIP/2.0 Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05 From: "4072930335" ;tag=as691eedde To: Contact: Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13 CSeq: 102 INVITE User-Agent: asterlink/1.2 Date: Sun, 09 Jul 2006 01:48:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 471 v=0 o=root 10036 10036 IN IP4 66.250.69.13 s=session c=IN IP4 66.250.69.13 t=0 0 m=audio 19082 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 - 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13 Sending to 66.250.69.13 : 5060 (non-NAT) Found peer 'asterlink-switch-03' 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 66.250.69.13:19082 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 66.250.69.13:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05;received=66.250.69.13 From: "4072930335" ;tag=as691eedde To: Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 66.250.69.13:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05;received=66.250.69.13 From: "4072930335" ;tag=as691eedde To: ;tag=as2a234c1b Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 8 21:50:03 WARNING[29936]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81577a8', 10 retries! aries*CLI> <-- SIP read from 66.250.69.13:5060: ACK sip:8006170141@host41.com SIP/2.0 Via: SIP/2.0/UDP 66.250.69.13:5060;branch=z9hG4bK6b164b05 From: "4072930335" ;tag=as691eedde To: ;tag=as2a234c1b Contact: Call-ID: 17c29dd64e0b98cb5faf961243c1b481@66.250.69.13 CSeq: 102 ACK User-Agent: asterlink/1.2 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '17c29dd64e0b98cb5faf961243c1b481@66.250.69.13'