Hi, I have setup and configured asterisk server using SIP. I defined two users 2000 and 2001 inthe sip.conf file. The extension defintions from sip.conf are shown below:
[2000] type=friend username=2000 secret=hi host=dynamic context=from-sip mailbox=100 [2001] type=friend username=2001 secret=hi host=dynamic context=from-sip mailbox=101 I made entries in extensions.conf, shown below, to place calls to 2000 and 2001 [from-sip] exten=>2000,1,Dial(SIP/2000,20) exten=>2000,2,Voicemail(u2000) exten=>2000,102,Voicemail(b2000) exten=>2000,103,Hangup exten=>2001,1,Dial(SIP/2001,20) exten=>2001,2,Voicemail(u2001) exten=>2001,102,Voicemail(b2001) exten=>2001,103,Hangup Initially I tested out the server by registering SJPhone user agents and successfully placing calls between them. Next I replaced the SJPhones with our VOIP gateways. Everytime I dialed either extension I always got the unavailable IVR message. I tried looking deeper into the problem and took ethereal traces and was able to isolate the problem. For some reason asterisk has problems in rewriting the TO header field when it forwards the INVITE request to the callee. This is what the TO header field looks like when it is sent by the caller to asterisk (192.168.0.44 is the IP address of asterisk): To: <sip:[EMAIL PROTECTED]:5060> and this is what it looks like when it is forwarded to the callee by asterisk (192.168.0.243 is the IP address of the callee). To: <sip:192.168.0.243> Since the URI does not contain the user part the callee replies with 404 not found and the call fails. I have thought hard, compared signaling traces but cant really make out how to make my gateways work. I would really appreciate any help. Regards, Danish _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users