You ok sir? Are you going to make it?
N.
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users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
>Sent: Wednesday, August 14, 2013 10:20 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] G729 Passthrough How To
>
>Anyone? :)
>
>N
users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
>Sent: Wednesday, August 14, 2013 10:20 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] G729 Passthrough How To
>
>Anyone? :)
>
>N
users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To
Not really no... And how do I make sure Aster
k-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: Re: [asterisk-users] G729 Passthrough How To
>
> Hey Eric, I do h
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.
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N
: Re: [asterisk-users] G729 Passthrough How To
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 wrote:
> I wanted to mention that I do not mind post
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 wrote:
> I wanted to mention that I do not mind posting the converted files on
> this list for future i
play
ringback to the caller.
-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 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
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
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
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.
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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 NEVER
, August 14, 2013 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To
Anyone? :)
N.
On 8/13/13, Nick Khamis wrote:
> Hello Everyone,
>
> We are currently experiencing some higher load on our servers, and
> sin
Anyone? :)
N.
On 8/13/13, Nick Khamis 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 the recording
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 the recording to the codec, and what about voicemail?
We are also using
Hello,
I'm doing g729 passthrough with asterisk 1.4 and it is working great whene i
call directly from my softphone to the destination number, but i'm not able
to do passthrough whene i make calls via Manager API Originate Command, the
calls always fail
_
hi group,
is there a way that SIP phones be allowed to use G.729 passthrough when
calling each other and when calling PSTN through Zap that asterisk
force the phones to use ulaw.
thanks,
ultor
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rcial Discussion
Assunto: Re: [Asterisk-Users] g729 passthrough?
i am trying to get G723 passthrough
get the same error.
how to configure passthrough for g723/g729 ?
On 4/24/05, Brian Capouch <[EMAIL PROTECTED]> wrote:
> jltaylor wrote:
>
> > ;;;
> >
I got some advice from Josh Colp that has helped with some of my problem:
it may have a little logic flaw in the way transcoding is supposed to be done, from
the way your message is I would say you are getting hit by this. (Upgrading to latest
CVS head will fix it)… but one solution is to be the
i am trying to get G723 passthrough
get the same error.
how to configure passthrough for g723/g729 ?
On 4/24/05, Brian Capouch <[EMAIL PROTECTED]> wrote:
> jltaylor wrote:
>
> > ;;;
> >
> > Brian,
> >
> > Add to the [general] section in sip.conf the following:
> >
> > di
jltaylor wrote:
;;;
Brian,
Add to the [general] section in sip.conf the following:
disallow=all
allow=g729
allow=ulaw
allow=alaw
For some reason Asterisk will not pass audio through itself without trying
to transcode unless you have this in your config.
Don't ask me why it will
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Capouch
Sent: Sunday, April 24, 2005 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] g729 passthrough?
I'm sitting here with my dunce cap on. My
I'm sitting here with my dunce cap on. My weak excuse is that I haven't
ever played with g729 before.
I have a Sipura 841. I have the phone config set to use g729. Its
appropriate sip.conf entry, and the IAX stanza for my ITSP all set to
disallow=all, allow=g729.
But as soon as I dial, I g
hello,
i wanted to use g729 in asterisk (iax to sip) as passthrough. Has anyone
got experience with configuring this or does someone know if this is
possible at all?
At the moment asterisk2 is always transcoding to alaw but but this
results in horrible voice quality.
phone1 behind NAT (sip) ->
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