[Asterisk-Users] How to use ASTCC with SIP ??

[Darren Wiebe]

I would recommend using the local trunk and then you just need a context 
that will dial out in your extensions.conf. Just put the context name 
into the "Peer/Trunk" field on the trunks page. Currently there is not 
support in astcc for oh-323. It would be trivial to add but....

Darren Wiebe
darren at aleph-com.net

Daniel Eboa wrote:

> Hello List,
>
> I’ve set up asterisk and install astcc application, everything was 
> well installed, but i’m having problem using astcc with SIP. I don’t 
> have any Trunk card or any other analogic VoIP card connected to my 
> asterisk box. I’m using SIP and asterisk-oh323 to connect to my VoIP 
> provider. Does anyone knows how I can use astcc to work with my config ?
>
> Thanks.
>
>------------------------------------------------------------------------
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>