Displaying 20 results from an estimated 10000 matches similar to: "Budgettone + G729"
2003 Sep 04
4
update re. Grandstream + SIP + Echo problems ..
well .. good news :)
i've just put in
txgain=1.0
rxgain=1.0
in my zapata.conf
and upgraded the Grandstream Budgettones i'm using to version 81
of the software and all seems fine .. there is still an echo but after
the first couple of seconds of call it vanishes, as the echocancelling
kicks in .. so far my client is happy :)
now .. i have one slight problem left .. although most of my
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
2003 Oct 16
7
I give up!!
i've just lost $2000 dollars or so on my first commercial asterisk
installation ..
i'm running a PIV class server, three Digium Wildcard FXO cards, and
10 Grandstream Budgettone SIP phones. The system was to be a PBX
for a small company. After over 2 months of pissing about, the client has
had his fill of asterisk problems, and asked me to take my equipment
out of the building. Obviously,
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?
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
-
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.
2005 May 27
3
G729 vs. gsm
I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm, G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?
I am interested in G729 because the internet in my
country is very expensive and I want to save every bit
possible. I want to use G729 because it takes less
bandwidth for
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 (
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
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
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 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
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 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
2004 Sep 29
1
Asterisk 1.00 Call quality problem
I upgraded from RC2 last night, but have a major call quality issue.
Heres our setup:
1 FXS and 1 FXO card.
Incoming/Outgoing calls via IAX trunking from our provider. G729
running between us and the VoIP provider.
Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2
SIP firmware.
Both these phones are using ULAW to the server, and we have plenty of
G729 licenses on the server.
2004 Dec 02
4
Codec Conversion
Hello,
Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything.
I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is
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
2003 Dec 24
8
G729 troubles
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip
2003 Jul 30
2
Call Transfer, Budgettone 100
hi,
can someone who has used Budgettone phones tell me how to do the
following:
an incoming call comes in and is answered by the receptionist.
she need to put the call on hold, speak to whoever the call is for,
and either (after that) pass on the call, otherwise speak again to
whoever was on the call and hang up ..
so far i've got as far as a blind transfer by pressing transfer button
and