Displaying 20 results from an estimated 3000 matches similar to: "g729"
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 =
2009 Aug 11
1
MixMonitor and Transcoding..
Can't find an answer to this, but maybe I've not looked hard enough ...
Does MixMonitor work without transcoding?
ie. if I have a g729 stream passing through and I'm recording it with
e.g. MixMonitor(/dump/filename.g729,b)
and specify g729 in the filename, does MixMonitor transcode both legs of
the stream to a format it can then "mix" then transcode it back to g729 to
2006 Apr 29
2
Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in place it works.
I use or used to use the licensed G729 Codec from Digium.
This is the error message from asterisk -vvg:
[app_playback.so] => (Sound File
2005 Apr 24
2
g729 passthrough?
I'm sitting here with my dunce cap on. My weak excuse is that I haven't
ever played with g729 before.
I have a Sipura 841. I have the phone config set to use g729. Its
appropriate sip.conf entry, and the IAX stanza for my ITSP all set to
disallow=all, allow=g729.
But as soon as I dial, I get a complaint from the server:
-- Call accepted by 66.225.202.72 (format g729)
--
2006 Jan 23
1
Installing the none commercial intel g729 codecs into Asterisk@Home 2.2?
Yep I did the same.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
2005 Aug 23
1
Can't get G729 working after buying a license.
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263)
when it should support g729 according to the config also listed below.
The real odd thing is I can place g729 calls to the router, just not
from the router to *. Anyone have any
2006 Jun 01
1
G729 + Native (files) MOH
Hello everyone,
One more little problem with a %100 g729 setup. Native moh:
musiconhold.conf:
[default]
mode=files
directory=/mnt/kd/moh/default
random=yes ; Play the files in a random order
ls /mnt/kd/moh/default
fpm-calm-river.g729 fpm-calm-river.ulaw fpm-sunshine.g729
fpm-sunshine.ulaw fpm-world-mix.g729 fpm-world-mix.ulaw
Place a call on hold:
Jun 1 14:55:30
2006 Jun 01
4
G729, voicemail, no codec_g729
I am trying to create a %100 g729 (with no transcoding) system (using a
Soekris, of course). I am running AstLinux with the native sounds, g729
is the only codec allowed, %100 SIP (g729 only allow=) - I think I am
covering all of my bases.
I have only "format=g729" in voicemail.conf. On an incoming call to a
mailbox, everything goes well until recording the message. When the
2019 Jul 05
2
Asterisk and Linphone
I have no speex translation
ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw
ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250 15000
alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000
2006 Jan 27
1
Installing the none commercial intel g729 codecsinto Asterisk@Home 2.2?
Thanks but this is for a test, I didn't buy the first one as it's a non commercial installation. I'm trying to test bandwidth etc so I need to try out how 4 of them handle the link simultaneously, I just don't know how to add a second test license.
Dean
________________________________
From: asterisk-users-bounces@lists.digium.com
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer <peername>" for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxxxxxx
* Name : 0049177xxxxxxx
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Record On feature : automon
2017 Nov 22
3
Chan Local, Originate and slin
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin
2017 Nov 22
2
Chan Local, Originate and slin
Again - when Originate is run from dialplan, i get:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin at 8000)->(slin at 192000)
ReadTranscode: No
When it's made with a call file (no matter how a call file is created), I
see
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
Please
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
2015 Sep 30
3
Change Asterisk MulticastRTP codec
Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information.
I have noticed that when I do a MULTICAST page and send data
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice
frame on Local/[removed number]@context-5c3e,2 of format ulaw since our
native format has changed to slin
Can anyone provide an English translation of what this means?
The extension is a Polycom IP 501
The only allowed formats are g.711u
MOH is MP3 files (obvious)
All prompts have been re-recorded in .ul uLaw
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
2006 Dec 28
1
1.4 - G729 - Have License - No path to translate from Zap to IAX2
Hello Everybody,
Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2:
[Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729)
[Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729
[Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing
[Dec 28 21:06:00] DEBUG[1756]
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
2014 Apr 29
1
"CBAnn" channel not going away in Asterisk 12
After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter
and then exit a conference room, I see:
-- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en')
-- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c>
-- Channel CBAnn/207-0000067f;2 left