asterisk users - Jan 2005 - How to use ASTCC with SIP ??

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On Jan 29, 2005, at 16:15, Daniel Eboa wrote:
> <image.tiff>?
> <image002.jpg>
Hi Daniel,

Would it be possible for you to turn off attaching two image files as 
signature replacements to each of your email and maybe use a text 
signature instead?

Thanks!

jens

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@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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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@lists.digium.com
[mailto:asterisk-users-bounces@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@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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>Asterisk-Users mailing list
>Asterisk-Users@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
>
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