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