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

[Romain Barrallon]

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)