for the info.
All the best,
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http://yziquel.homelinux.org/
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Miguel Molina a écrit :
Guillaume Yziquel escribió:
So what is this permission issue? Where are the changes from 1.0 to
1.1 documented?
When I was testing asterisk 1.6.0.X with the AMI Originate action, I
fell into the same issue as you. I found that it was that the
permissions now
Message: Permission denied
So what is this permission issue? Where are the changes from 1.0 to 1.1
documented?
All the best,
--
Guillaume Yziquel
http://yziquel.homelinux.org/
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gives me 'Ringing' but sometimes it only gives
me 'Ring', which drives my application to 100% CPU (OK, not a very
resilient app for now...)
My question is: is 'Ring' a specific state for a channel, or is it a
known bug in the AMI?
All the best,
--
Guillaume Yziquel
http
in this setup.
Is it possible to do such a thing with Asterisk? Does it need really
special tweaking of Asterisk conf files?
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Guillaume Yziquel
http://yziquel.homelinux.org/
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work
without any trouble
Thank you for this valuable information.
--
Guillaume Yziquel
http://yziquel.homelinux.org/
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
the 'A' in the ACL field of the 'sip show
peers' command might be.
I've been unable to find this information on the net. Help would be
appreciated.
--
Guillaume Yziquel
http://yziquel.homelinux.org/
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asynchronous
integration, and guidance would be appreciated.
All the best,
--
Guillaume Yziquel
http://yziquel.homelinux.org/
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,
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Guillaume Yziquel
http://yziquel.homelinux.org/
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Steve Howes a écrit :
On 4 Sep 2009, at 10:36, Guillaume Yziquel wrote:
Uniqueid: asterisk-1252055630.26702
I'm really wondering how the Uniqueid works. Why is it incremented? What
is the dot for in the Uniqueid?
The bit before the dot is a unix timestamp (Fri, 04 Sep 2009 09:13:50
GMT
Call from asterisk and not from 'yziquel'. Moreover when I
pick up the phone it says You are now talking to asterisk, and then
Zoiper closes the call immediately.
There's surely something I do not get right here, and I'd appreciate
some help.
All the best,
Guillaume YZiquel
priorities) in 1 context. =-
Thanks a lot.
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http://yziquel.homelinux.org/
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be
the most common reasons behind this? Please feel free to ask for more
relevant details.
All the best,
Guillaume Yziquel.
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Guillaume Yziquel a écrit :
Hello.
I've been seting up a small VoIP setup, with roughly 5 persons, doing
essentially some Meetme conferences.
People have been experiencing some quality problems with the sound.
Essentially delay, and some tolerable echo.
I'd appreciate advice on how
, or to documentation on
this issue? Does this sound like a NAT issue?
All the best,
Guillaume Yziquel.
Here's the beginning of the sip.conf file:
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default
this issue?
All the best,
Guillaume Yziquel.
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Steve Totaro a écrit :
On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel
guillaume.yziq...@citycable.ch wrote:
Hello.
I've set up and configured an Asterisk server to make SIP phone calls to
external classic phones.
However, it happens that after 15 or 30 seconds, the phone call drops
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