Hi!
Is this possible to make asterisk follow the dial plan after executing
VoicemailMain?
Thanks,
Michel
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Hi!
I try to make my Asterisk contact a SIP user thanks to a redirect
server. In fact Asterisk try to reach a SIP address that is redirected
to the good one.
The error response is:
*CLI> -- Executing Dial("OSS/dsp", "sip/[EMAIL PROTECTED]|30|
H|g") in new stack
-- Called [EMAIL PROTECTED
Ok.Thanks a lot! I will try!
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- Message d'origine
De: Mojo with Horan & Company, LLC <[EMAIL PROTECTED]>
A: Zenone <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
Discussion
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 22:39
> Yes, on a Zap FXO channel, w
- Message d'origine
De: Eric ManxPower Wieling <[EMAIL PROTECTED]>
A: Zenone <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
Discussion
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 12:54
> Zenone wrote:
> > But my quest
But my question was, is it possible to free the channel if it rings too
long?
Michel
Message sent using UebiMiau 2.7.8
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Hi!
Does a ringing timer exist in asterisk to control ringing duration? If not,
is there a way to control ringing duration?
Thanks in advance for your help,
Michel
Message sent using UebiMiau 2.7.8
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