|
Hi list. I’m having problems handling the incoming calls when
using asterisk as a sip client. For example, on sip.conf I have lines like this: register => user1:[EMAIL PROTECTED]/777 register => user2:[EMAIL PROTECTED]/888 [host1] context=hostnumber1 type=friend insecure=very disallow=all allow=alaw allow=ulaw canreinvite=no secret=password1 username=user1 host=123.321.123.321 fromuser=user1 fromdomain=123.321.123.321 nat=no language=es dtmfmode=rfc2833 [host2] context=hostnumber2 type=friend insecure=very disallow=all allow=alaw allow=ulaw canreinvite=no secret=password2 username=user2 host=123.321.123.321 fromuser=user2 fromdomain=123.321.123.321 nat=no language=es dtmfmode=rfc2833 Then, under extensions.conf I have this: [hostnumber1] exten => 777,1,Echo() [hostnumber2] exten => 888,1,SayDigits(888) I gues this is the right way to send incoming calls for
user1 to an echo test, and incoming for user2 to get some digits spoken. I tried this simple thing yesterday, but Asterisk responds
“Not Found” to the INVITE sent from outside. Please, someone tell me to which context in extensions.conf
will those calls be tried to match. Thanks for your time ;) Bye. Alejandro Mejia |
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
