Asterisk might be unable to transcode rtp type from downstream to upstream,
or vice versa.
There's a bug reported here, for asterisk 12 or above, using chan_sip.
https://issues.asterisk.org/jira/browse/ASTERISK-25676
It says that you could avoid the bug by using chan_pjsip, but you still
encounter
Hi Matt
Thanks for your response. I have tried with two GXV3175 with same result.
Let me dig deep on this to find out the route cause
Sam
Matthew Jordan wrote:
On Thu, Jun 13, 2013 at 12:04 PM, resea...@businesstz.com wrote:
Hi there
I have asterisk 10.11.1 which seems to have problem
On Thu, Jun 13, 2013 at 12:04 PM, resea...@businesstz.com wrote:
Hi there
I have asterisk 10.11.1 which seems to have problem negotiating codec.
Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p
and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw,
- Original Message -
From: Ryan McGuire rdmcguir...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 3, 2011 9:47:42 AM
Subject: Re: [asterisk-users] Codec negotiation issue (no audio format found
to offer)
From looking into this, it appears as if this is due
, David Vossel dvos...@digium.com wrote:
- Original Message -
From: Ryan McGuire rdmcguir...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 3, 2011 9:47:42 AM
Subject: Re: [asterisk-users] Codec negotiation issue (no audio format
found to offer)
From looking
, August 3, 2011 9:47:42 AM
Subject: Re: [asterisk-users] Codec negotiation issue (no audio format
found to offer)
From looking into this, it appears as if this is due to Asterisk
negotiating the legs separately as if they were not related to the
same call. So the ingress leg negotiates
From looking into this, it appears as if this is due to Asterisk negotiating
the legs separately as if they were not related to the same call. So the
ingress leg negotiates ulaw, and despite it knowing that the peer also
supports g729 fails the call since it's already decided on ulaw and the
Hi,
If you will send call without answering on asterisk and have directrtpsetup=yes
in sip.conf codec negociation will always be between UAs so any matched codec
will work fine. If you are answering call on asterisk then dialing it out to
next UA then you need to add canreinvite=yes for both
On 08/03/2010 04:21 PM, Philipp von Klitzing wrote:
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
I have the following setting in
Only when I configure my Grandstream to use only G726 (I have 8
choices), I see that the g726-codec is used.
When I configure 7 x g726 and 1 x alaw, then again alaw is used !
Is it normal that Asterisk has such a great preference for alaw ?! The
moment the peer suggests codec alaw (even if
Hi!
Question 1 :
[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
why is combined alaw|g726 and not g726|alaw (reverse) ??
Guess: Here the order presented has no meaning for the order of codec
Hello Philipp,
thank you for your answer.
On 08/03/2010 01:21 PM, Philipp von Klitzing wrote:
Question 3 :
How can I get g726 as first preferred codec ??
Which Asterisk version are you using?
Using Asterisk 1.4.30
* check if you have disallow/allow settings in the [general]
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
For quick testing to see if the codec works at all: Configure your phones
to do g726 only (so
Hi!
In the [general] section of sip.conf I have :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
So change the order there and see what happens.
* look at the variable SIP_CODEC for the inbound (first) call leg, and
in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
On 26 June 2010 22:08, Ryan Wagoner rswago...@gmail.com wrote:
I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
between the phones using g722. However Asterisk is negotiating g722
for calls going out my
...@lists.digium.com] On Behalf Of Steve Davies
Sent: Tuesday, June 29, 2010 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec negotiation
On 26 June 2010 22:08, Ryan Wagoner rswago...@gmail.com wrote:
I have Polycom phones that support the g722 codec
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec negotiation
On 26 June 2010 22:08, Ryan Wagoner rswago...@gmail.com wrote:
I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
I have reported a codec-issue, but there is no solution. Will this patch
also answer my question ??
https://issues.asterisk.org/view.php?id=17020
Jonas.
On 06/29/2010 09:42 PM, Mindaugas Kezys wrote:
Try this:
Hi!
Because the codec is already chosen before the call is made, and you
told it that g722 is permitted.
There are all sorts of discussions in play about codec negotiation,
but at this point in time, if you want different behaviour you'll need to
go and code it yourself
Look at the list
Hi!
Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
Most probably - who on this list would you like to test it for you? ;-
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
Because the codec is already chosen before the call is made, and you
told it that g722 is permitted.
There are all sorts of discussions in play about codec negotiation,
but at this point
If my memory serves me right, we could use Thomson ST2030 and Asterisk 1.4.
Have you tried with another soft or hardphone ?
2008/7/7 Vinz486 [EMAIL PROTECTED]:
Hi all,
i'm trouble with codec setup on an asterisk machine 1.4.18 and some
Thomson ST2030 as extensions.
In the users.conf file
On Mon, Jul 7, 2008 at 12:18 PM, Olivier [EMAIL PROTECTED] wrote:
If my memory serves me right, we could use Thomson ST2030 and Asterisk 1.4.
Have you tried with another soft or hardphone ?
Why not???
___
-- Bandwidth and Colocation Provided by
I do not need g723.1 codec, this is not the problem, here is another
description of the problem:
The client offer 2 codecs (g729 and g723) for all calls, my server accept
only g729, so normally the client server will negotiate the codec and both
sides agrees on g729, but this does not happened
So who do you pay to use the G723 codec?
Best regards,
Al Bochter
http://www.BochterServices.com
---
See what we are selling at auction
http://www.epier.com/auctions.asp?bochterservices
On 7/12/07, O. Kamal [EMAIL PROTECTED] wrote:
I am having a problem with my asterisk gateway, it is accepting only G729,
the client is offering G729 and G723.1, however for some reasons, around
15% of calls are rejected due to failed codec negotiation giving an codec
error No compatible codecs,
On Thu, 2007-07-12 at 14:39 -0400, Al Bochter wrote:
So who do you pay to use the G723 codec?
It's possible to use the G.723.1 codec with Asterisk by buying a Digium
TC400B transcoder card[1]. Without that card, the best Asterisk can do
is to pass through the packets, but it can't doing any
- Douglas Garstang [EMAIL PROTECTED] wrote:
I expected Asterisk to send G711 instead, as that's what is set in
[general] in sip.conf
And as you've already learned, Asterisk will reorder the codecs in the outbound
INVITE so that the codec used on the incoming channel is listed as first
On Fri July 21 2006 18:33, Woodoo People .pGa!
[EMAIL PROTECTED] wrote:
don't forget the following:
if canreinvite=yes, asterisk will NOT stay in mediapath, so, it going to
ask both parties to negotiate codec, and say hello to the stream. (if
both parties supports g729, and can negotiate it,
I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first
preference, and the phone on the fast link to use G711 as it's first
preference.
sip.conf has:
[general]
allow=ulaw
allow=g729
[slow-link] ;
No, we aren't intending to check for available g729 codecs
that's why we wanted to have ulaw as a backup when no g729 codecs
where available.
That won't work. If it's trying to use G729, it will still try even
when the licenses are all in use. So you need to either force it g729
Just an idea:
Put this Slow-Phone sip account into sip realtime database, and
outside of asterisk manage to verify G729 licenses availability and
script it to your SIP-realtime.
This way every call to this SIP account will go to SIP realtime
database that is being changed by an external script
On Jul 21, 2006, at 3:01 AM, Woodoo People .pGa! wrote:
No, we aren't intending to check for available g729 codecs
that's why we wanted to have ulaw as a backup when no g729 codecs
where available.
That won't work. If it's trying to use G729, it will still try even
when the licenses are
]
Sent: Friday, July 21, 2006 4:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
Just an idea:
Put this Slow-Phone sip account into sip realtime database, and
outside of asterisk manage to verify G729 licenses availability
Douglas Garstang wrote:
Can't put it in a realtime database. We have multiple Asterisk boxes in a
cluster, and it's a well known fact that multiple Asterisk boxes using realime
cannot query a common MySQL database. Sounds crazy, but true.
You spread some amazing well-known facts on this
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
Douglas Garstang wrote:
Can't put it in a realtime database. We have
Douglas Garstang wrote:
Would you like me to dig up the posts from Keving Fleming stating that this is
known not to work Brian?
As I recall those posts have to do with the way your particular setup
required ARA to work with a failover/redundant cluster system you were
building.
Beyond
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
Douglas Garstang wrote:
Would you like me to dig up
I must agree,
we use 2 Ser in front of 4 asterisk sharing the same database cluster.
Olivier
Brian Capouch a écrit :
Douglas Garstang wrote:
Would you like me to dig up the posts from Keving Fleming stating
that this is known not to work Brian?
As I recall those posts have to do with
Well, I wish someone would tell Kevin Fleming that.
-Original Message-
From: olivier.taylor [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
I must agree
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk
Users Mailing List - Non-Commercial Discussion
[mailto:[EMAIL PROTECTED]
Sent: Fri, 21 Jul 2006 16:21:15
-0300
Subject: RE: [asterisk-users] Codec Negotiation
Well, I wish someone would
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Codec Negotiation
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL
On Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote: I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use G729, and one on a fast link where I'd like to use ulaw. My sip.conf has: [general]
DiscussionSubject: Re: [asterisk-users] Codec
Negotiation
On Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote:
I'm a little confused about Asterisk codec negotiation. Hopefully
someone can help.
I
have two phones, one on a slow link where I'd like to use G729
On Jul 20, 2006, at 11:00 AM, Douglas Garstang wrote:Subject: Re: [asterisk-users] Codec NegotiationOn Jul 20, 2006, at 10:16 AM, Douglas Garstang wrote:I'm a little confused about Asterisk codec negotiation. Hopefully someone can help. I have two phones, one on a slow link where I'd like to use
Message-From: Martin Joseph
[mailto:[EMAIL PROTECTED]Sent: Thursday, July 20, 2006 12:34
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Codec
Negotiation
On Jul 20, 2006, at 11:00 AM, Douglas Garstang wrote:
Subject: Re
, 2006 12:34 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Codec NegotiationOn Jul 20, 2006, at 11:00 AM, Douglas Garstang wrote:Subject: Re: [asterisk-users] Codec NegotiationOn Jul 20, 2006, at 10:16 AM, Douglas
On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote:
I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first
preference, and the phone on the fast link to use G711 as it's first
preference.
sip.conf has:
-Original Message-
From: Martin Joseph [mailto:[EMAIL PROTECTED]
Sent: Monday, July 17, 2006 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
On Jul 17, 2006, at 9:25 AM, Douglas Garstang wrote:
I have two
Hi,
Ronald Voermans wrote:
I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've
configured two incoming phonenumbers. One phonenumber is for
voice-calls, the other one for receiving faxes. I want the incoming
voice-calls to be coded by the G.729 codec, and the fax-number by
---
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Florian Overkamp
Verzonden: donderdag 9 februari 2006 18:27
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Codec negotiation
Hi,
Ronald Voermans wrote
Hi Ronald,
Ronald Voermans wrote:
What exactly do you mean by seperating traffic in to differt SIP peers?
The situation is as follows:
I have OpenSer connected to our SIP provider/PSTN Provider (the answer
to your question: Enertel).
Ah 'kay.
Asterisk registers to OpenSer, which then
:[EMAIL PROTECTED] Namens Florian Overkamp
Verzonden: donderdag 9 februari 2006 23:38
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Codec negotiation
Hi Ronald,
Ronald Voermans wrote:
What exactly do you mean by seperating traffic in to differt SIP
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Salim
Sent: dinsdag 25 januari 2005 22:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Codec negotiation
The order matters in asterisk so if you want GSM to take priority over
G729,
simply put
for example) the current CVS-HEAD version doesn't
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Salim
Sent: dinsdag 25 januari 2005 22:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Codec negotiation
On Tue, 2005-01-25 at 16:10 -0500, Mohammed Salim wrote:
The order matters in asterisk so if you want GSM to take priority over G729,
simply put that ahead of the G729... so your settings should be:
Allow=all
This would allow everything so the next lines would be redundant.
Disallow=all
The codec is selected by asterisk depending upon the codecs that you
have allowed for the particular channel context and your setting of the
bandwidth= parameter.
It would be nice if you could set things up so that an inbound call
could force * to a higher bandwidth codec when needed (for
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Eissler
Sent: Tuesday, January 25, 2005 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Codec negotiation
The codec is selected by asterisk depending upon the codecs that you
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Eissler
Sent: Tuesday, January 25, 2005 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Codec negotiation
The codec is selected by asterisk depending upon the codecs
On Sat, 2004-11-20 at 18:48 +0100, Tamas J wrote:
Hello!
I would like to know wether it is possible to have end-to-end codec
negotiation in iax2?
What I mean is...
In case the user dials a number available through PSTN, let's force to
use alaw (the client is in LAN) to overcome unneeded
Why do you need 729? I just called your IAXTel number using GSM and
connected fine.
Michael
On Wed, 18 Feb 2004 08:29:48 +0100, dkwok wrote:
I have outgoing connection to iaxtel and another iax server A.
iax server A only accept g729 codec while iaxtel is something I am not
quite sure of. At
I think your problem comes from a misunderstanding of how the calls are
placed. With your canreinvite=no in the ATA section, you end up with the
ATA negotiating with asterisk for a call leg. Then you have asterisk
negotiating for the other call leg. Since the RTP stream is going
through asterisk,
Try using canreinvite=yes in all three contexts. Would that screw-up ATA, I
do not know, cause I have no Cisco's ?
SW
Message: 5
Date: Mon, 05 Jan 2004 02:29:49 -0500
From: SamW [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Codec Negotiation Does not seem to work as
expected
Thanks for all who is helping.
I tried, canreinvite=yes on all contexts but that do not seem to work as
well. But the issue is not related to negotiating between end points,
but for me, asterisk do not have a proper configuration scheme which
works, to the requirement of the user. The
Title: Re: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
Steve,
My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about
Hello Eduardo,
Wednesday, December 17, 2003, 1:08:00 AM, you wrote:
EG Hi list,
EG I'm with a little problem on codec negotiation between a cisco827 and
EG asterisk.
EG My sip.conf is like that:
EG [general]
EG port = 5060
EG bindaddr = 0.0.0.0
EG context = default
EG
- Original Message -
From: Eduardo Goncalves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 16, 2003 1:08 PM
Subject: [Asterisk-Users] codec negotiation
Hi list,
I'm with a little problem on codec negotiation between a cisco827 and
asterisk.
My sip.conf is like
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