B.H.
But if the final response is 480 doesn't it mean that the call was placed
but there was no reply?
On Aug 13, 2013 10:30 PM, Shishir Pokharel shishir.pokha...@on24.com
wrote:
*21.1.5* http://tools.ietf.org/html/rfc3261#section-21.1.5* 183
Session Progress*
** **
** **
The 183
2013/8/13 Administrator TOOTAI ad...@tootai.net
Le 13/08/2013 16:41, Olivier a écrit :
Hello,
Hi
[...]
How can I work around this ?
Suggestions ?
Answer the call, wait few seconds and then ring Bobs extension. If
asterisk detect fax it already sended to fax extension so Bobs
hi,
i have strange problem with call-limit/groupcount limiting. i set up
limit of 2 calls.
i'm using both methods but a for few times i have problem with
successfull fraud with more calls than 2
asterisk is 1.8.22
someone with the same problem?
any ideas how to solve or debug this problem?
Anyone? :)
N.
On 8/13/13, Nick Khamis sym...@gmail.com wrote:
Hello Everyone,
We are currently experiencing some higher load on our servers, and
since signaling comes into our servers on G729, we would like to
implement G729 pass-through. A few questions arise, do we need to
convert all
If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to
transcode. This means all calls must use only g729, sound files must be in
g729 format and no early audio, inband ringing or anything else which might
cause Asterisk to require a temp transcoding path.
In my
Hey!!! Eric thank you so much for your response. Could you guys please
direct us in achieving as much as possible. For example:
* What linux command can we use to convert all recording to G729
* Which files do we need to convert and there locations
* For *testing* how do we make sure Asterisk
I forgot to mention that all our equipment (phones etc..) are using
G729, and this is for internal use over the net. The problem,
concurrent calls, and bad bandwidth at some locations...
N.
--
_
-- Bandwidth and Colocation
As my understanding Asterisk always pass-thu g729 if both ends have this
codec.
But if you answer the call or play some audio before dialing to end point
then asterisk stay between both legs.
In case of VM. you should install g729 if your prompts are in g729 format.
As a2billing play voice prompts
Dne 14.8.2013 13:35, Marek Cervenka napsal(a):
hi,
i have strange problem with call-limit/groupcount limiting. i set up
limit of 2 calls.
i'm using both methods but a for few times i have problem with
successfull fraud with more calls than 2
asterisk is 1.8.22
someone with the same
Hello Ashgar,
Thank you so much for your response. As removing A2B is not an option
we would first like to begin by converting all audio files (Asterisk,
VM, A2B prompts etc...) to G729 to minimize unneeded trascoding. Linux
commands and the list of recording would be a great help. Sorry, not
new
file convert in the Asterisk CLI, IF you have the g729 codec installed.
You need to convert every single file you may play to a caller
You can't force Asterisk to never attempt transcoding, the most you can do is
force all sip.conf entries to use g729. It will still transcode to play
ringback
Hey Eric, I do have the codec installed, and I remember hearing about
the CLI command to convert. Is there a recent how-to of blog already
discussing this somewhere?
N.
On 8/14/13, Nick Khamis sym...@gmail.com wrote:
I wanted to mention that I do not mind posting the converted files on
this
I have no idea, though Google might.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I wanted to mention that I do not mind posting the converted files on
this list for future individuals, given that I am not doing anything
illegal...
N.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Not really no... And how do I make sure Asterisk always generates
prompts and VM recordings in G729 from now on. This is also hard to
find information..
N.
On 8/14/13, Eric Wieling ewiel...@nyigc.com wrote:
I have no idea, though Google might.
-Original Message-
From:
Asterisk does not generate prompts. You force G729 in VM by only allowing
g729 in voicemail.conf.Try reading the Asterisk book.
Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at
http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is
I have a system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card
using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't
relevant to the question).
With DAHDI and Asterisk started, the system appears to run normally, as
far as I can tell from limited testing.
I am monitoring
Hi list, I'm trying to attach a Avaya with Asterisk, call the extension 3241
to 1042 belonging to avaya, but only sounds rings and when I pick up the
phone keeps ringing
08-14-13 07:26:17 AM-856ms Line = 18, Channel = 1, SIP Message = Response,
Direction = From Switch, From = 3241@172.16.8.40,
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