Displaying 20 results from an estimated 10000 matches similar to: "Managing codecs"
2004 Jun 24
2
How to force G729
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that:
[mypstngate]
type=friend
2008 Jul 07
2
Codec negotiation for Thomson ST2030 and g729
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 for internal extension i have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Without any codec installed (i mean with original g729 of asterisk)
all go fine, calling from an extension to one other:
Peer User/ANR Call ID Seq (Tx/Rx) Format
2004 Dec 16
1
Dynamically Choose Codec for Bandwidth Management
Is there any way to set Asterisk to choose what codec to allow for a new
call based on current usage? In other words... be able to define a max
number of ulaw calls, then after that only allowing g729? The idea here is
that in general, a T-1 should be enough for our offices to have phone +
citrix + some video (got good QoS in place already). But for usage spikes,
user experience would be kept
2004 Dec 20
7
One SIP peer use 2 diff codecs?
I asked this question once before with no answer. Hopefully someone can
help me as I cannot see a way to do this. I am wanting to differentiate
inbound calls voice from FAX. The purpose of course voice gets g729 and
FAX gets 711 (ulaw). The problem I'm having is everytime it matches the
SIP peer (like it should) but it's always goes to the prefered codec.
Anyone have suggestions on how to
2003 Oct 20
1
Setvar SIP_CODEC
Hello,
I have
a couple of 7960 and a quad T1 card on my asterisk box. I want to let
the phones to use g729 when they "talk" to each other, but to use g711
when I'm going to route the call out of my network using the T1 card.
Everything works just fine between the phones, but in order to be able
to make calls through T1 I have to disallow the g729.
For this purpose I have the
2008 Mar 10
1
Local music on hold -- mohinterpret=passthrough assymetrical ?
Hi list,
I'm planning and testing a distributed asterisk deployment
throughout several sites; each will be connected to the PSTN
and all of them among themselves via IAX trunks. Phones
will be SIP.
I guess I already "solved" (worked-around, actually) asterisk's
codec negotiation limitations regarding local G.711 utilization vs.
remote G.729 while minimizing
2005 Aug 06
0
g729 pass-thru for sip provider and g711 ulaw for conference and voicemail
Hello,
I'd like to use g729 pass-thru when I dial out to a sip provider from my
IP phone but because I have no license for g729 I'd like to use g711 ulaw
for asterisk voicemail, conference bridge and other services.
When I set in [general] section of sip.conf the following:
disalow=all
allow=g729
allow=ulaw
the g279 pass-thru works fine with my SIP provider but
when I call the
2009 Oct 20
1
Is there a way to force a codec on an incoming sip uri call?
Hello,
I'd like to implement some public sip uri's that poeple can call into
and get an echo test. Is there a way to force a codec so that users
can test various codecs?
Something like:
echo-test at example.com (negotiates whatever codec, is there a way to
figure out what codec was negotiated and tell the user)
echo-test-g711 at example.com (forces g711)
echo-test-g729 at
2013 Jan 17
1
g729 codec over SIP Trunk between CCM and Asterisk
Hello,
My problem is, outgoing calls (from asterisk to CCM) work fine but incoming
(from CCM to Asterisk) does not work because of CCM is trying to use g729
over SIP trunk. I have found that link after a quick search. Problem is the
same as in link below (However my Asterisk version is 1.8.13) and solution
seems to have H323 trunk between CCM and Asterisk for using g729 codec. The
post was
2009 Feb 25
1
SIP_CODEC variable
Hi,
I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec
being used. I have exclusively Polycom phones for this test, and basically I
want all communications to use g729 (preferred codec), except for pagine 20
phones (which busts my g729 license count). In that case I want to use gsm.
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate
2020 Sep 24
2
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2009 Jan 16
0
No subject
FYI, not sure if it's of use to you... but... The digium tc400b is a transcoder card that can offload upto 120 channels of transcoding for g729 <-> ulaw... It's available as PCI only, but, if that's OK, it could be an alternative to replacing your server... G729 licenses are not needed when using that card...
There have been posts by some people about having multiple CPU
2010 Aug 02
6
Codec negotiation : expecting G726, getting G711a (alaw)
Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 & 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 & 4 x G729.
The SIP peers are both defined as :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
This is the
2009 Jan 16
0
No subject
potentially starting to see problems...
FYI, not sure if it's of use to you... but... The digium tc400b is a
transcoder card that can offload upto 120 channels of transcoding for g729
<-> ulaw... It's available as PCI only, but, if that's OK, it could be an
alternative to replacing your server... G729 licenses are not needed when
using that card...
There have been posts by some
2008 Oct 23
0
command - set sip_codec- does not work with asterisk-1.4.21
hello:
i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729.
I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2
2004 Jul 26
1
voicemail+g729
HI ALL;
I found in the following page:
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
1-If I could record all IVR promts in G729 format
2-If I could record voicemail in g279 format with """format_g729.c"""""
then I donot need any g729 license (I suppose all my clients have g729 ip phones)
My question is, how
2011 Mar 06
1
Early codec selection / negotiation
Hi,
This seems to be a fairly common question, but I have Googled for this quite
a bit and looked at the Asterisk documentation/book and haven't been able to
find an answer.
My question is:
Can I get my IP phone to select a different codec depending on the final
destination of each call?
I've got these things connected to my Asterisk box:
- Snom 300 phone (supports g729 and
2004 Sep 05
5
Asterisk Conferencing using g729
Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW.
My iax.conf file includes the following under the general section
2004 Jun 24
6
R: How to force G729
>> allow=ulaw
>Why don't you remove this?
Because I need some other users to be able to call out using ULAW over the same PSTN gateway...
-Manuel
___________________________________________________
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
2009 May 08
0
G279 install in 1.6.0.9 ? [SOLVED]
2009/5/8 Olivier <oza-4h07 at myamail.com>
> Hello,
>
> Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are
> instructions to install G729 software.
> (I think I followed instructions step by step but g729 license doesn't seem
> to show up).
>
> My question is :
> Is the command bellow still up to date ?
>
> >g729 show
I suddenly