Displaying 20 results from an estimated 10000 matches similar to: "G729 - Part cut"
2004 Apr 20
1
h323 and oh323 g711 to g729 please help
Hello list,
I have many IP hardphones like Siemens 300 basic ( old ) , cisco
ata.. etc
I need: G711 from old phones must be convert to G729 via asterisk and
send to provider ( G729 from digium )
I have this problems:
oh323 (last version):
-------------
asterisk work with this driver ok for old phones, if I only
faststart=no . But problem with codec , asterisk can speak with
provider (
2007 Sep 26
2
My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is G729 pass-through from a gateway to a phone. I think
I even have transcoding working, which makes me more confused on
what's wrong with my pass-through. It must be a configuration issue.
The basics...
*CLI> core show version
Asterisk
2005 Mar 28
1
H323: g711-g729 transcoding
I have a connect to * via H.323/g711 from device A and want to connect
to B which want for H.323/g729
h323.conf contains
disallow=all
allow=alaw
allow=g729
but outgoing faststart/TCS contains only g711 (from h323_request(format)
i think) and so no codec negotiation and no voice.
Howto run up g711/H323 -> * -> g729/H323
PS intel's g729 was used. ast 1.0.3-6
PPS
stupid
-
2004 Apr 18
1
h323 oh323 g729 please help !
Hello list,
I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc
I need: G711 from old phones must be convert to G729 via asterisk and send to provider
I have this problem:
oh323 (last version):
-------------
asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable
2007 Sep 14
1
g729 on 1.4.10.1
I have a fresh 1.4.10.1 installation that appears to have a problem
with g729 pass-through. I can see the gateway in question sending an
INVITE using g729b. However, the Asterisk is only sending g711 in the
INVITE to my Polycom phone.
[gateway]
disallow=all
allow=g729
[phone]
disallow=all
allow=ulaw
allow=alaw
allow=g729
There's the codec configs for the gateway and the phone in question.
2004 Jun 18
2
C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 -> g729 -> asterisk -> g711 -> C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?
2003 Oct 03
1
Budgettone + G729
hi there ..
I asked sometime ago regarding getting a Budgettone
working with Asterisk over G729.
My system is quite simple, Asterisk server with 1 G 729 license
installed, and 10 Grandstream phones. Only one of them needs
G729, because it's on a remote link via an ADSL bridge. The
rest run happily on G711 on a local network.
I added the lines
disallow=all
allow=g729
to the sip.conf entry
2006 Nov 20
2
Recording g729
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Helvetica, Arial, sans-serif">Before ordering I want to be
2008 Apr 01
1
g729 encoder/decoder
How does the g729 encoder/decoder count in regards to the total number
of licenses and how does it count an encoder/decoder? I looked on the
wiki and don't really see anything that explains it. In other words,
how do the calls below count (assume no reinvite)?
g729 phone calls into voicemail
g729 phone calls g711 phone
g729 phone calls other g729 phone
2007 Jan 04
2
Dimensioning a 50 sip phone installation
Hi,
Some help with dimensioning the server will be gladly accepted.
-50 sip phones (g729) or g711(to avoid transcoding) in LAN
-an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN
-Some sporadic conferencing with no more than 2 sip phones and maybe 2
or 3 calls coming from the E1 for a total of 5 people in a conference.
The asterisk server will get an E1(pri) via one
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
2020 Sep 22
2
Negotiates g729 but RTP contains g711
Hi,
We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip
2006 Feb 20
1
g729 quality at GSM bitrates
Greetings all,
I'm trying to improve the codec selection on a few of the asterisk boxes we
have to keep the g729 licences free for calls from ATAs that don't support
anything apart from g711 and g729. GSM seems to offer noticably inferior
call quality (at least when using a softphone + decent headphones), but it's
about where I want the bitrate to be.
I know there are lots of Speex
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
2010 Mar 17
3
SIP codec negotiation / manipulation
We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second
2007 Nov 28
4
G729/MOH Quality
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not attempted to use much else so far.
I've ruled out timing issues on the system generating the MOH itself
(ztdummy on the PBX itself, our Zaptel gateway
2011 Jun 13
5
No audio after a reinvite changing codec
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:
UAC ASTERISK UAS | ASTERISK UAC
PROVIDER
g711 <----------------------> g711
2007 Feb 14
6
Fax with T.38
Hi all,
I install the last version of Asterisk and I tried to send faxes, but
nothing works.
Here is my configuration:
Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA
<----> Analog Fax 2
I tried Analog Fax 2 -> Analog Fax but nothing works!!
In the Patton configuration I put G711 and no silence suppression.
In asterisk I have
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if
sip-server request g711)
I have 2 SIP-services to
2006 Feb 17
4
one way / irratic voice over iax and g729
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier
The first Asterisk system is running 1.2 and the second is running 1.0.
When using g726 from the handset all the way thru to Asterisk2(then 729
for the carrier leg) calls go thru fine, but when using g729, there