hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!!<br><a href="mailto:asterisk-users@lists.digium.com"></a><br><br>--- (13 headers 16 lines) ---<br>
Sending to AA.BBB.CCC.DD : 28127 (NAT)<br>Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk.<br>Found user '701' for '701'<br>Found RTP audio format 107<br>Found RTP audio format 0<br>
Found RTP audio format 8<br>Found RTP audio format 18<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://192.168.0.73:40958">192.168.0.73:40958</a><br>Found unknown media description format BV32 for ID 107<br>
Found audio description format G729 for ID 18<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Peer audio RTP is at port <a href="http://192.168.0.73:40958">192.168.0.73:40958</a><br>Looking for 702 in from-internal (domain <a href="http://ABC.dyndns.org">ABC.dyndns.org</a>)<br>
list_route: hop: <sip:701@123.456.789.000:37587><br>acerdebian*CLI><br><--- Transmitting (NAT) to 123.456.789.000:28127 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.0.73:15158;branch=z9hG4bK-d8754z-0a540c5d3439c271-1---d8754z-;received=123.456.789.000;rport=28127<br>
From: "me"<<a href="mailto:sip%3A701@ABC.dyndns.org">sip:701@ABC.dyndns.org</a>>;tag=3c08d834<br>To: "702"<<a href="mailto:sip%3A702@ABC.dyndns.org">sip:702@ABC.dyndns.org</a>><br>Call-ID: Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk.<br>
CSeq: 2 INVITE<br>User-Agent: Asterisk PBX 1.6.0.6<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A702@192.168.1.4">sip:702@192.168.1.4</a>><br>
Content-Length: 0<br><br><br><------------><br> -- Executing [702@from-internal:1] ResetCDR("SIP/701-0864f1b8", "") in new stack<br> -- Executing [702@from-internal:2] NoCDR("SIP/701-0864f1b8", "") in new stack<br>
-- Executing [702@from-internal:3] Wait("SIP/701-0864f1b8", "1") in new stack<br>Retransmitting #1 (NAT) to 123.456.789.000:9855:<br>OPTIONS sip:701@123.456.789.000:37587;rinstance=9428b8620cd7a907 SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK782c5851;rport<br>Max-Forwards: 70<br>From: "Unknown" <<a href="mailto:sip%3AUnknown@192.168.1.4">sip:Unknown@192.168.1.4</a>>;tag=as43db5836<br>To: <sip:701@123.456.789.000:37587;rinstance=9428b8620cd7a907><br>
Contact: <<a href="mailto:sip%3AUnknown@192.168.1.4">sip:Unknown@192.168.1.4</a>><br>Call-ID: <a href="mailto:564e1f392a3b289f3b655a7200a67378@192.168.1.4">564e1f392a3b289f3b655a7200a67378@192.168.1.4</a><br>CSeq: 102 OPTIONS<br>
User-Agent: Asterisk PBX 1.6.0.6<br>Date: Thu, 02 Jul 2009 18:51:58 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br><br>