Hi everyone,
I'm having some errors using Asterisk for incoming/outgoing calls. I
have SER working with mysql without problems, all my internal users autenticate
at SER and then if any number begins with a "1", Ser forwards the call to
asterisk. Asterisk takes the forward and act as a Sip Client to make the call.
I need to do that because My VoIP numbers are at Go2call, and I need asterisk
to authenticate at go2call server to take line signal, to complete
incoming/outgoing calls. At this moment none of them are working. When I try to
dial number 13125899691 I receive the following Error:
.
.
.
Looking for 13125899691 in OUTGOING
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
.
.
.
I followed the tutorial at the following sites:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
http://www.voip-info.org/wiki-Asterisk+SIP+channels
http://www.voip-info.org/wiki-Asterisk+introduction
Has anybody have this Scenario, that could give me a hand ? It's really
starting to piss me off ...
Thanks in Advance
Best Regards.
--
Felipe Martins
Mundivox Communications
Tecnologia e Projetos
[EMAIL PROTECTED]
Tel.: +55 +21 +3820 8839
Cel.: +55 +21 +9823 8602
Fax.: +55 +21 +3820 8844
www.mundivox.com
_______________________________________________
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
