Romain Barrallon
2005-May-26 13:34 UTC
[Asterisk-Users] Asterisk con X-lite : Register Ok but no calls (404 Not found)
Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer. I can't understand why asterisk doesn't found the users if they are registred... It's making a "Scheduling Call Destruction". My config files are : sip.conf : [general]>>context=default ; Default context for incoming calls >>recordhistory=yes ; Record SIP history by default >>port=5060 ; UDP Port to bind to (SIP standard port is 5060) >>bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) >>srvlookup=yes >> >>[1111] >>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! >>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed >>type=friend >>username=1111 >>secret=1111 >>callerid="Thibaud" <1111> >>host=dynamic >>context=from-sip >>allow=ulaw >>qualify=yes >> >>[2222] >>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! >>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed >>type=friend >>username=2222 >>secret=2222 >>callerid="Florentin" <2222> >>host=dynamic >>context=from-sip >>allow=ulaw >>qualify=yesextensions.conf :>>[bogon-calls >>exten => _.,1,Congestion >> >>[from-sip] >> >>exten => 1111,1,Dial(SIP/1111,20) >>exten => 1111,2,Voicemail(u1111) >>exten => 1111,102,Voicemail(b1111) >>exten => 1111,103,Hangup >> >>exten => 2222,1,Dial(SIP/2222,20) >>exten => 2222,2,Voicemail(u2222) >>exten => 2222,102,Voicemail(b2222) >>exten => 2222,103,Hagup >> >>exten => 9999,1,VoicemailMain(${CALLERIDNUM})The critical SIP exchange is : SEND TIME: 440651449 SEND >> *.*.*.173:5060 INVITE sip:2222@*.*.*.173 SIP/2.0 Via: SIP/2.0/UDP *.*.*.172:5060;rport;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049 From: Asterisk <sip:1111@*.*.*.173>;tag=93980267 To: <sip:2222@*.*.*.173> Contact: <sip:1111@*.*.*.172:5060> Call-ID: 7FB284DC-20B7-8A06-F426-2E514014A6AA@*.*.*.172 CSeq: 30470 INVITE Proxy-Authorization: Digest username="1111",realm="asterisk",nonce="2c887956",response="79eb7583cec4b45e867189dfa7d515dd",uri="sip:2222@200.1.27.173" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 285 v=0 o=1111 440651420 440651437 IN IP4 *.*.*.172 s=X-Lite c=IN IP4 *.*.*.172 t=0 0 m=audio 10000 RTP/AVP 0 8 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv RECEIVE TIME: 440651467 RECEIVE << *.*.*.173:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP *.*.*.172:5060;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049 From: Asterisk <sip:1111@*.*.*.173>;tag=93980267 To: <sip:2222@*.*.*.173>;tag=as6c9ced81 Call-ID: 7FB284DC-20B7-8A06-F426-2E514014A6AA@*.*.*.172 CSeq: 30470 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2222@*.*.*.173> Content-Length: 0 -- Romain Barrallon - Etudiant en T?l?communications, Services et Usages ? l'INSA de Lyon (France) - Estudiante de intercambio en la Universidad Tecnica Federico Santa Maria de Valpara?so (Chile)
Romain Barrallon
2005-May-27 14:27 UTC
[Asterisk-Users] Asterisk con X-lite : Register Ok but no calls (404 Not found)
Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer. I can't understand why asterisk doesn't found the users if they are registred... It's making a "Scheduling Call Destruction". My config files are : sip.conf : [general]>>context=default ; Default context for incoming calls >>recordhistory=yes ; Record SIP history by default >>port=5060 ; UDP Port to bind to (SIP standard port is 5060) >>bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) >>srvlookup=yes >> >>[1111] >>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! >>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed >>type=friend >>username=1111 >>secret=1111 >>callerid="Thibaud" <1111> >>host=dynamic >>context=from-sip >>allow=ulaw >>qualify=yes >> >>[2222] >>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! >>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed >>type=friend >>username=2222 >>secret=2222 >>callerid="Florentin" <2222> >>host=dynamic >>context=from-sip >>allow=ulaw >>qualify=yesextensions.conf :>>[bogon-calls >>exten => _.,1,Congestion >> >>[from-sip] >> >>exten => 1111,1,Dial(SIP/1111,20) >>exten => 1111,2,Voicemail(u1111) >>exten => 1111,102,Voicemail(b1111) >>exten => 1111,103,Hangup >> >>exten => 2222,1,Dial(SIP/2222,20) >>exten => 2222,2,Voicemail(u2222) >>exten => 2222,102,Voicemail(b2222) >>exten => 2222,103,Hagup >> >>exten => 9999,1,VoicemailMain(${CALLERIDNUM})The critical SIP exchange is : SEND TIME: 440651449 SEND >> *.*.*.173:5060 INVITE sip:2222@*.*.*.173 SIP/2.0 Via: SIP/2.0/UDP *.*.*.172:5060;rport;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049 From: Asterisk <sip:1111@*.*.*.173>;tag=93980267 To: <sip:2222@*.*.*.173> Contact: <sip:1111@*.*.*.172:5060> Call-ID: 7FB284DC-20B7-8A06-F426-2E514014A6AA@*.*.*.172 CSeq: 30470 INVITE Proxy-Authorization: Digest username="1111",realm="asterisk",nonce="2c887956",response="79eb7583cec4b45e867189dfa7d515dd",uri="sip:2222@200.1.27.173" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 285 v=0 o=1111 440651420 440651437 IN IP4 *.*.*.172 s=X-Lite c=IN IP4 *.*.*.172 t=0 0 m=audio 10000 RTP/AVP 0 8 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv RECEIVE TIME: 440651467 RECEIVE << *.*.*.173:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP *.*.*.172:5060;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049 From: Asterisk <sip:1111@*.*.*.173>;tag=93980267 To: <sip:2222@*.*.*.173>;tag=as6c9ced81 Call-ID: 7FB284DC-20B7-8A06-F426-2E514014A6AA@*.*.*.172 CSeq: 30470 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2222@*.*.*.173> Content-Length: 0 -- Romain Barrallon - Etudiant en T?l?communications, Services et Usages ? l'INSA de Lyon (France) - Estudiante de intercambio en la Universidad Tecnica Federico Santa Maria de Valpara?so (Chile)