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

[Karl H. Putz]

The current astcc Makefile puts the sound files into the wrong directory.
It uses /usr/share/asterisk/sounds but it should be
/var/lib/asterisk/sounds.


Karl Putz

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Daniel Eboa
> Sent: Saturday, January 29, 2005 12:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] How to use ASTCC with SIP ??
>
>
>
> I got this error when i try to dial:
>
> -- Executing Answer("SIP/8000104-71a3", "") in new stack
>     -- Executing DeadAGI("SIP/8000104-71a3", "astcc.agi") in new stack
>     -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
> astcc-tone does not exist in any format
> Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
> astcc-accountnum does not exist in any format
> Jan 29 18:11:37 WARNING[3412]: file.c:779 ast_streamfile: Unable to open
> astcc-accountnum (format alaw): No such file or directory
>   == Spawn extension (prepaid, 77, 2) exited non-zero on
> 'SIP/8000104-71a3'
>
> Can somebody tell me why and how to solve it ??
>
> Regards.
> Daniel.
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Darren
> Wiebe
> Sent: samedi 29 janvier 2005 18:12
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How to use ASTCC with SIP ??
>
> 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.
> >
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