Displaying 20 results from an estimated 20000 matches similar to: "music on hold file formats"
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 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
2008 Feb 04
6
transcoder
Dears
Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to transcode the incoming sip calls from g711 to g723 or
ilbc or g729 ..... and forward it to a media gateway ..
Regards
Khaled chehab
*********************************************
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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
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 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
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 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.
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
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?
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
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
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
2004 Jul 27
2
g729 + GSM + g723
Folks!
We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found.
Here is the config I have used:
-------------------------------
Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2
User1 is in USA on Broadband Cable
User2 is in India on 64Kbps ISDN Line
User1 using SIPURA SPA 2000
user2 using Xten professsional(X-pro)
2010 Aug 05
2
AMI Command
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
2008 Jul 01
3
music on hold realtime
Hi,
Is it possible to use realtime for Music On Hold?
Is it also possible to store the music/audio files on the database, same
way a voicemail can be stored on the database?
Thank You
Regards,
Nhadie
2006 Mar 03
2
Asterisk Fax Question
Hi All,
I want to configure fax with Asterisk and I found that we can do this reliably
using G711 codec only. Currently my provider is supporting G729 and G711.
During the call initiation the call starts with G729 (1'st priority) and
somehow if the receiver is unable to receive call then we are providing the
Caller to send a fax, but at that point they are using G729 codec. At this
point how
2007 Jun 06
4
Best Codec
We are evaluating starting a small VoIP consumer based platform.
What is the best codec to use with customers using primarily DSL as
internet connectivity?
I know that g729 is the king-all, but I want to know what the rest of
the professional are using out there. g729 has a cost involved, so does
the cost really offset the performance? Or is it better to go with g711
to start off?
We plan
2006 Jan 13
2
ILBC to G711 transcoding experince ?
Hello All,
Anyone here has experience of accepting a ilbc call and sending it on g711 or g729
I am having problem in VOICE , call goes though but there is no voice.
Senario:
Call is coming in from Machine A to Machine B, sending to Machine C
Machine B is an asterisk box, transcoding it from IBLC to G711 and g729.
Problem:
Voice is not appearing on the sip user sitting on machine A
Already
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