2012-09-07 16:13, David M. Lee skrev:
On Sep 7, 2012, at 1:49 AM, Johan Wilfer wrote:
Hi!
I'm trying to limit the permissions for a AMI-account. But I'm a little bit confused by
the permissions. The commands I use are (output from manager show commands,
btw: privilege col seems cropped?):
Hello,
We want to use Asterisk as a proxy to translate between Skinny/SCCP and
SIP, with as little as possible work required in between.
Does Asterisk have a way for custom programs to read and write raw packets?
If we can get the input data in a readable format and output it in the
required
Hi Danny,
I am running it asynchronously because adhearsion needs it.
Regards,
Pavel
Date: Wed, 5 Sep 2012 11:28:46 -0500
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Async AGI
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Hi,
I have 10 agents who are pstn lines in queue and would like that when they
answer the rtp should go directly
Is it at all possible in queues?
If yes what could be bothering it from happening?
Thanks,
Israel
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Hi list,
since I enabled call recording in my ConfBridge bridge profile, after
about every 5th held conference (sometimes after 3, sometimes after 10,
it depends), when the last users exits the conference, Asterisk crashes.
Is this a known error? I'm using Asterisk 10.7.1 on CentOS 64bit with
That being the case, you are going to need to do a dialplan modification
something like this:
Exten = _X.,1.Set(answerstatus=false)
exten = _X.,2,AGI(agi:async)
exten = _X.,3,Gotoif($[${answerstatus}==false?7:4)
exten = _X.,4,Answer
exten = _X.,5,Playback(some-message)
exten =
- Original Message -
From: Markus unive...@truemetal.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 10, 2012 8:19:47 AM
Subject: [asterisk-users] Asterisk crashing when recording ConfBridge calls
(10.7.1)
On Sep 10, 2012, at 2:38 AM, Johan Wilfer wrote:
Thank you David for the feedback.
I reported the following bugs:
https://issues.asterisk.org/jira/browse/ASTERISK-20397 (all bug)
https://issues.asterisk.org/jira/browse/ASTERISK-20396 (cropped col)
Thanks!
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David M. Lee
Digium, Inc. |
I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43
(I doubt it's very version-specific). I won't have hands on the kit until the
end of the week, but I have listened to some recordings. It doesn't happen on
every call - only sometimes.
Basically, it is a call from one
Danny as I mentioned in the first message the dialplan execution stops
after AGI(agi:async). So the provided example doesn't help.
Regards,
Pavel
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New to
Perhaps the AGI is improperly terminating and should stop and return an
error instead of just dying like it does now. Just catch the failing
condition and return a variable to the dialplan so life will be good.
From: asterisk-users-boun...@lists.digium.com
Greetings list,
I've seen a few errors recently in our logs along the lines of:
[Sep 10 17:41:41] WARNING[6719] app_voicemail.c: Save failed. Not
moving message: destination folder full.
maxmsg in voicemail.conf is set to 1000.
I've checked the mailboxes on the server in question, and the
2012-09-10 18:13, Tony Mountifield skrev:
I'm trying to diagnose potential causes of an issue with MixMonitor in 1.4.43
(I doubt it's very version-specific). I won't have hands on the kit until the
end of the week, but I have listened to some recordings. It doesn't happen on
every call - only
On 10/9/12 6:48 pm, Danny Nicholas wrote:
What flavor of asterisk? Realtime or just files? Post your voicemail.conf.
Flat files, latest 1.4.x
Kind regards,
Chris
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On Sep 10, 2012, at 11:16 AM, Pavel Siderov wrote:
Danny as I mentioned in the first message the dialplan execution stops after
AGI(agi:async). So the provided example doesn't help.
Regards,
Pavel
Pavel,
This may be due to the asynchronous nature of Async AGI. The AGI(agi:async)
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