Displaying 20 results from an estimated 6000 matches similar to: "Codec Selection in asterisk"
2004 Oct 04
1
Will there be any support for iLBC in IAXClients soon?
Hello Folks,
I noticed that all of the IaxClient based softphones with exception of
Firefly only seem to have support for GSM but what about iLBC?
The quality is excellent with iLBC even on a dialup connection! Meanwhile
while the audio on GSM often sounds scratchy. Is anyone looking to
implement iLBC in an IaxClient based softphone soon?
Errol
Biz4Web Solutions Limited
2007 Apr 27
4
Unable to find a codec translation path from ilbc to ulaw
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:
SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway
I get the following error:
"Unable to find a codec translation path from ilbc to ulaw"
Setup SIP-phone:
disallow=all
2004 Oct 01
2
Forcing a codec
Hi,
I'm having trouble explicitly forcing a codec between sip devices. Am
I missing something or is this not really possible?
I have a grandstream registering to asterisk, named sip0. Sip0 registers,
via sip, to another asterisk box, sip1. When I place a call from the
grandstream, it will travel through sip0 to sip1, where it is then placed
to the PSTN. Nothing can reinvite, this path is
2004 Nov 09
2
ssh login
sorry - hope this question is not tooo silly,
but i needed to "autologin" to a remote machine
found out that this works fine for me:
sftp -opassword=PASSWORD USER at 192.168.1.1 << EOF
cd ANYDIRECTORY
get ANYFILE
bye
EOF
why isn't this (sftp -opassword=SECRET USER at 192.168.1.1),
setting the password with -opassword=PASSWORD, documentated
anywhere?
bug or feature?
kind
2005 Feb 16
3
HELP!!!!!!!!
Hi,
I have installed two X-Lite phones and they're able to login successfully.
The two phones plus the Asterisk system are all on the same LAN with private
addresses assigned to each of them. When a call is initiated and is picked
up on the other end, there is completely no sound at all (as in the line
goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and
SPX.
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
2005 Jun 09
1
IAX2 Max Retries dropped calls Firefly
Hi
We've recently set up and are using with success 1.0.7 using a Junghanns
quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works
very well, however we're getting cases where sometimes the call just drops.
>From setting more verbose modes we get a log which is shown below. The problem
seems to be the maxretries message which comes from chan_iax2. We are using
2005 May 13
3
Audio quality
I'm a new Asterisk user. I've managed to set it up to do everything I
want except sound good. Currently, Asterisk sounds considerably worse
than my cell phone. I know VOIP can be _better_ than my cell phone,
because I've heard Skype do it. (Using 32k iLBC, I believe.)
I did an experiment with audio quality:
1) I made a recording which was pretty good. I used an iSight
2008 Aug 09
1
how to know what codec is being used
Hi,
how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all.
i unset all codecs on x-lite except ilbc.
i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does
2007 May 04
2
Asterisk Codec Translation Table
Hello list,
I have always though codec translation table is dircetly connected to system speed, utill i came across this:
in my lab, i have 2 boxes,
First box is an Intel Celeron 1.7 GHZ with 256M RAM:
show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw
2015 Nov 09
2
availability of target type state within a dumpxml result
Hi,
I'm trying to find out what is the minimum release I need to use to have this field 'state' available ?
<channel type='unix'>
<source mode='bind' path='//var/lib/libvirt/qemu/dummy_agent'/>
<target type='virtio' name=dummy' state='connected'/>
<alias name='channel0'/>
<address
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
2004 Nov 18
3
Is H323 dying?
Hello,
I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems)
After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering:
Does asterisk consider H323 so achaic that it does
2004 Dec 04
1
Codec translator problem (g723.1,ilbc => alaw)
Hi, I cannot get SIP channel working with folowing codec configuration:
[sip]
disallow=all
allow=g723.1 ;I need this codec between sip phones (BT100)
allow=ilbc ;Use this codec to others
Calling between BT100 SIP phones is OK - asterisk makes native bridge
(with g723.1) between them.
When I'm calling from SIP to other channel (iax,zap,...), asterisk is
not able to chose right codec
2005 Mar 21
1
iLBC codec and mute issues
I tried using the iLBC codec, and whlie I like it, I ran into a
strange issue. I did a few searches on Google and haven't found anyone
with the same issue as this.
Anyhow, I was using a Plantronics analog headset and box plugged into
a Digium TDM card, dialed out over my VoIP provider's IAX channel to
the PSTN.
I was in a conference call which is running on an Avaya PBX (which
2008 Jul 29
4
xfs on 5.2 (live cd + dvd)
Hello,
I'm planning a server migration and being able to mount xfs file systems
with the live cd would be a cruical feature.
So before I download and try ... can anyone tell me whether the xfs is
included in the 5.2 live cd?
Later on I'm planning to install a new system with xen, 3ware 9550SX-4LP
and xfs. The xen domains are of course located on xfs partitions.
Do these features come
2005 Jun 10
3
Comparison
I'm not an expert either, but I see people choosing iLBC over speex
all the time with asterisk; partly it's because they have more
market share in hardphones, and partly it's because of marketing and
such. (another reason is that iLBC source is included in asterisk,
and speex is only compiled in if you have the speex development stuff
on your machine when you compile
2005 Mar 11
1
NuFone Configuration [problem]
Hello,
I am trying to configure the my asterisk box here with the following
**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xxxxxx
***extensions.conf:***
exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN}
I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan.
2003 Apr 16
4
iLBC
i tried asterisk ilbc codec against kphone. when the call got
connected, i started to immediately get these kind of message to the
console:
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)?
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of
2004 May 19
1
using iLBC
I want use iLBC and have following in mind, please help me is it possible ?
ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM
SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM
1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC
because a lack of codec).
2. SIP incoming codec should be iLBC (snom is ALAW).
3. SIP outgoing codec should be iLBC /snom