[Asterisk-Users] Asterisk con X-lite : Register Ok but no calls (404 Not found)

[Zoa]

They find the users but not the extension.

Have a look at
http://www.asteriskguru.com/tutorials/xlite_softphone.html for a
complete configuration guide.
http://www.asteriskguru.com/tutorials/softphones.html is the first page.

zoa


Romain Barrallon wrote:

>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=yes
>>>
>>>
>
>extensions.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 at 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 Télécommunications, Services et Usages à l'INSA de Lyon (France)
>- Estudiante de intercambio en la Universidad Tecnica Federico Santa
>Maria de Valparaíso (Chile)
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