Displaying 20 results from an estimated 10000 matches similar to: "3way calling / codec problem"
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
2007 Jun 22
1
Does Early Media have to be Ulaw?
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid="Test hone 1" <+19256002182>
host=dynamic
canreinvite=no
secret=password
context=test
disallow=all
allow=g729
[level3]
type=peer
host=xxx.yyy.16.99
context=default
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
2005 Jul 20
1
Zap channel(s), meetme and codecs/licences
Hi all,
Some simple questions about codecs:
What codec does the Zap channel use by default?
Can this default be changed, and to what? (g729 too?)
What codec does meetme use? (I think this is ulaw, but asking to be sure)
Can you use another codec, or does everything have to be transcoded to ulaw?
Finally ... if I have a 3way call going, between 1 g729 caller and two
other callers, do I need one
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
2007 Jun 28
2
Fax passthrough howto codec upspeed
Hello everybody,
Just was wondering if somebody can help for G711 fax passthrough w/ asterisk.
The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax
When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT
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
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
2006 Apr 19
1
Codec problem from SIP to H323
Hello.
I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:
- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my asterisk doesn't need to "transcode"
(I don't have licences for g729)
- sip.conf
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
2006 Dec 15
2
call from h323 to SIP
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP
address for this machine
allow=all
extension.conf:
exten = 3298,1,Answer
exten =
2004 Nov 24
2
Codec control
How can i control the codec for the calls. For example I have 3 SIP
phones registered to asterisk
The firs two are in the local area network (behind nat)- I want to use
g711 between them and to connect directly (canreinvite=yes)
and the third is in internet - want all calls to it and from it to use
g729 and media to go through asterisk.
So if Phone 1 calls Phone 2 the codec to be g711, but when
2013 Oct 01
2
is g729 codec free? or under license???
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when i
run "core show codecs" in asterisk, there is a g729 codec in the list so i
assume that i can use it for my channels. but connection can not be set
when i use it for my h323 channel.
i read somewhere that codec g729 is a commercial codec and i should buy its
license in order to use it. is it true? if
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
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
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
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
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP
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