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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Alejandro Imass
Sent: Tuesday, 23 November 2010 6:18 a.m.
To:
On Mon, Nov 22, 2010 at 11:46:21PM +0100, antselva wrote:
I have a problem with dialing status.
I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls.
When I call fixed telephone (not mobile phone) after few ringing the
status change to answer but the phone is still ringing, so if I
I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta
or release candidate ?
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Thanks, Phil
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Hi all,
I would like to know if something I am trying to do is possible.
I currently using 1.8 and would like to make recordings in wideband
(16khz)
I have an aastra 6739i which supports the g722 codec.
I made an agi application in which I would like an user to record a
promptlist.
I
Hello,
I want to analyze the asterisk logs files, looking for all kind of
errors, ¿Anyboby knows any asterisk logs analyzer?
Thanks all,
Voipcrazy
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New
Hi,
Please don't consider this as an advertizing, but since we received feedback
from the community that the only voice quality assessment solutions
available are worth 50K, we have started to develop an Asterisk based voice
quality monitoring framework. The core technology for perceptual
On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote:
I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta
or release candidate ?
It will be Asterisk 1.8.1-rc1 and that is now available (as of a few minutes
ago)
http://www.asterisk.org/node/51466
Leif.
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The Asterisk Development Team has announced the first release candidate of
Asterisk 1.8.1. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.1-rc1 resolves several issues reported by the
community and
On 10-11-23 08:24 AM, Henry Dogger wrote:
I have an aastra 6739i which supports the g722 codec.
Which format setting do I need to be able to record in wideband?
Tried: wav, gsm, pcm. Nothing seems to give me the result I desire.
Shouldn't you try g722 as the format?
Leif.
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I tried this also, but now I can't listen to the prompts on my pc.
On the phone it seems fine, but how do I know it's wideband.
G722 should be 16 khz 14 bit.
I use goldwave to listen to the prompts, but this does not support such
a setting.
Can someone point me in the right direction perhaps?
On Tue, Nov 23, 2010 at 10:58 AM, Henry Dogger h.dog...@telecats.nl wrote:
I tried this also, but now I can't listen to the prompts on my pc.
On the phone it seems fine, but how do I know it's wideband.
G722 should be 16 khz 14 bit.
I use goldwave to listen to the prompts, but this does not
- Original Message -
On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote:
I have read the wiki entry but unsure when we would likely see a
1.8.0.1 beta or release candidate ?
It will be Asterisk 1.8.1-rc1 and that is now available (as of a few
minutes ago)
Can someone show me how to patch the new asterisk 1.8.1-rc1 code with the sip
transfer fix described @ https://issues.asterisk.org/view.php?id=18185 ?
I tried 'svn co -r 295866
http://svn.asterisk.org/svn/asterisk/trunk/[apps|include|main]', but the files
I
checked out didnt seem to be ones
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
us...@lists.digium.com) commented about Re: [asterisk-users] Someone has
hacked
into our :
On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
I have the log now. I'd like to know
Dear Daniel,
Thank you very much for your support.
What you write is correct, after disabling the early-connect option the
Patton 4554 started to work as desired.
Il 23/11/2010 12.05, Daniel Tryba ha scritto:
On Mon, Nov 22, 2010 at 11:46:21PM +0100, antselva wrote:
I have a problem
On 21 November 2010 23:13, Jonathan Hunter jmhunt...@gmail.com wrote:
I've been experiencing trouble with my DAHDI channels for some time and
have finally decided to try and resolve the issue.
Essentially, the problem I am having is that when a call comes in, and my
DAHDI phones therefore
Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesherasterisk-
us...@lists.digium.com) commented about Re: [asterisk-users] Someone has
hacked
into our :
On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
I
Hello,
I'm trying to use SIP_HEADER function on my dialplan but I receive this
message (on the console):
pbx.c:3367 ast_func_read: Function SIP_Header not registered
Why?
Thank's
- Bakko
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On 11/23/10 14:18, Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
us...@lists.digium.com) commented about Re: [asterisk-users] Someone has
hacked
into our :
On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
On Tue, 23 Nov 2010 18:57:16 -0500
bakko asannu...@gmail.com wrote:
Hello,
I'm trying to use SIP_HEADER function on my dialplan but I receive
this message (on the console):
pbx.c:3367 ast_func_read: Function SIP_Header not registered
Why?
I believe function names are case sensitive,
Jonathan Hunter wrote:
On 21 November 2010 23:13, Jonathan Hunter jmhunt...@gmail.com
mailto:jmhunt...@gmail.com wrote:
I've been experiencing trouble with my DAHDI channels for some
time and have finally decided to try and resolve the issue.
Essentially, the problem I am
Hi Chad,
thank you very much,
now work...
Best Regards
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please reply on this if u know
On 11/18/2010 09:24 AM, Nikhil wrote:
Hi everyone
Anyone please explain me How Account code is use for billing.,
Thanks
Nikhil
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Reposting my answer:
It's often used to signify an account that a call is associated with,
and I believe that was the original intent.
You can use it, however, to signify any important data about a call,
such as the pin number used to access a certain call feature (like
FreePBX allows you to do
Hi all,
Perhaps someone has dealt with it before.
I want to activate a bunch of my own scripts after someone has registred
om my asterisk, or when his cient has de-registerded.
have been skimming through AGI and AMI, and seen a lot of nice features,
but not the (de-)registering events.
Kind
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