Hello people, I am having some issues with my new SIP provider. The sip provider gives me only an IP address to configure my sip account, since they do allow by IP address and not by username password. This all configuration appears to work well, since I can originate a call and it will ring the destination, and I can originate a call from PSTN and * will "see" it. But none of both call difections will be stabilished. If I originate a call from * to a PSTN number, with a sip debug I get: Destroying call '086247827f2ab1c83b4f39a5389b17bc@200.59.45.210' pbx*CLI> <-- SIP read from 200.123.190.50:5060: SIP/2.0 500 Server Internal Error To: <sip:44242904851@200.123.190.50>;tag=3364745030-621025 From: "CrossFone" <sip:Unknown@200.59.45.210>;tag=as4d4398b9 Call-ID: 67d7050e602585775316c78b17d9ea4a@200.59.45.210 CSeq: 102 INVITE Contact: sip:44242904851@200.123.190.50:5060 Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 500 "Server Internal Error" back from 200.123.190.50 Transmitting (no NAT) to 200.123.190.50:5060: ACK sip:44242904851@200.123.190.50 SIP/2.0 Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport From: "CrossFone" <sip:Unknown@200.59.45.210>;tag=as4d4398b9 To: <sip:44242904851@200.123.190.50>;tag=3364745030-621025 Contact: <sip:Unknown@200.59.45.210> Call-ID: 67d7050e602585775316c78b17d9ea4a@200.59.45.210 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/CrossFone-087b3e40 is circuit-busy Destroying call '67d7050e602585775316c78b17d9ea4a@200.59.45.210' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/1501-087acbf8", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/1501-087acbf8", "Dial failed due to CONGESTION") in new stack -- Executing Macro("SIP/1501-087acbf8", "outisbusy|") in new stack -- Executing Playback("SIP/1501-087acbf8", "all-circuits-busy-now") in new stack If I make a call to my SIP number, it will ring till I pickup the phone, when I pickup the phone, I get: <-- SIP read from 201.216.206.221:62477: --- (0 headers 1 lines)--- pbx*CLI> <-- SIP read from 200.123.190.50:5060: INVITE sip:1159174200@200.59.45.210 SIP/2.0 Max-Forwards: 70 Session-Expires: 3600;Refresher=uac Supported: timer To: 1159174200 <sip:1159174200@200.123.190.50> From: <sip:1152184829@200.123.190.50:5060>;tag=3364745421-27664 Call-ID: 53757-3364745421-27638@msw1.subnet32.net CSeq: 1 INVITE Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe Contact: sip:1152184829@200.123.190.50:5060 Content-Type: application/sdp Content-Length: 170 v=0 o=NexTone-MSW 1234 0 IN IP4 200.123.190.53 s=sip call c=IN IP4 200.123.190.53 t=0 0 m=audio 21660 RTP/AVP 18 4 4 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes --- (12 headers 8 lines)--- Using INVITE request as basis request - 53757-3364745421-27638@msw1.subnet32.net Sending to 200.123.190.50 : 5060 (non-NAT) Found peer 'CrossFone' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 0 Peer audio RTP is at port 200.123.190.53:21660 Found description format G729 Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x105 (g723|ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 1159174200 in from-sip-external (domain 200.59.45.210) list_route: hop: <sip:1152184829@200.123.190.50:5060> Transmitting (no NAT) to 200.123.190.50:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123 .190.50 From: <sip:1152184829@200.123.190.50:5060>;tag=3364745421-27664 To: 1159174200 <sip:1159174200@200.123.190.50> Call-ID: 53757-3364745421-27638@msw1.subnet32.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1159174200@200.59.45.210> Content-Length: 0 --- -- Executing NoOp("SIP/5060-087ace18", "Received incoming SIP connection from unknown peer to 1159174200") in new stack -- Executing Set("SIP/5060-087ace18", "DID=1159174200") in new stack -- Executing Goto("SIP/5060-087ace18", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/5060-087ace18", "0?from-trunk|1159174200|1") in new stack -- Executing Set("SIP/5060-087ace18", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-08-16 19:27:24 UTC. -- Executing Answer("SIP/5060-087ace18", "") in new stack We're at 200.59.45.210 port 19920 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 200.123.190.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123 .190.50 From: <sip:1152184829@200.123.190.50:5060>;tag=3364745421-27664 To: 1159174200 <sip:1159174200@200.123.190.50>;tag=as7635cbf2 Call-ID: 53757-3364745421-27638@msw1.subnet32.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1159174200@200.59.45.210> Content-Type: application/sdp Content-Length: 231 v=0 o=root 2815 2815 IN IP4 200.59.45.210 s=session c=IN IP4 200.59.45.210 t=0 0 m=audio 19920 RTP/AVP 18 4 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/5060-087ace18", "2") in new stack pbx*CLI> <-- SIP read from 200.123.190.50:5060: ACK sip:1159174200@200.59.45.210 SIP/2.0 Max-Forwards: 70 To: 1159174200 <sip:1159174200@200.123.190.50>;tag=as7635cbf2 From: <sip:1152184829@200.123.190.50:5060>;tag=3364745421-27664 Call-ID: 53757-3364745421-27638@msw1.subnet32.net CSeq: 1 ACK Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe Contact: sip:1152184829@200.123.190.50:5060 Content-Length: 0 --- (9 headers 0 lines)--- pbx*CLI> <-- SIP read from 201.216.206.221:62364: any idea?