On 28/02/2013, at 6:08 PM, Richard Kenner ken...@gnat.com wrote:
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
Do you have transcode_via_sln set in asterisk.conf?
No, but as I said in a later email, I found the problem: when computing the
cost of a path, any downconvert has the same cost. So
siren14 - slin - slin32
is the same cost as
siren14 - slin16 - slin32
which is wrong.
I fixed this
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
seems odd that there's no transcoder directly to slin32 since anything
else
I'm quite fond of GSM610 as a low(ish) bandwidth codec - although it isn't as
good as (say) speex or Silk,
it is widely supported, and European users have had years of cellphone use to
get used to the specific
sound of a GSM call. So you can often go from a GSM610 supporting handset all
the
Is it a good idea to use asterisk transcoding from G711 to iLBC or should I
find out any other solution not involving transcoding (f.e. using G.729 that is
supported in both sides). I'm worried about voice quality and trying to avoid
paying for G.729 licensing.
Anybody with experience or
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
Is it a good idea to use asterisk transcoding from G711 to iLBC or
should I find out any other solution not involving transcoding (f.e.
using G.729 that is supported in both sides). I'm worried about voice
quality and trying to avoid paying
On 04/15/2012 07:26 PM, Patrick Lists wrote:
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
Is it a good idea to use asterisk transcoding from G711 to iLBC or
should I find out any other solution not involving transcoding (f.e.
using G.729 that is supported in both sides). I'm worried
: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: segunda-feira, 22 de março de 2010 2:33
To: Asterisk User MailList
Subject: [asterisk-users] Transcoding question
We are getting ready to install a client that uses g729 when
de março de 2010 2:33
To: Asterisk User MailList
Subject: [asterisk-users] Transcoding question
We are getting ready to install a client that uses g729 when talking to
their SIP provider to minimize bandwidth usage. We are going to want to
be able to record the calls using AMI monitor actions
To: Asterisk User MailList
Subject: [asterisk-users] Transcoding question
We are getting ready to install a client that uses g729 when talking to
their SIP provider to minimize bandwidth usage. We are going to want to
be able to record the calls using AMI monitor actions into wav sound
files. All
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: segunda-feira, 22 de março de 2010 2:33
To: Asterisk User MailList
Subject: [asterisk-users] Transcoding question
We are getting ready
We are getting ready to install a client that uses g729 when talking to their
SIP provider to minimize bandwidth usage. We are going to want to be able to
record the calls using AMI monitor actions into wav sound files. All the phones
are soft phone running on Windows XP systems.
Questions I
-users] Transcoding question
We are getting ready to install a client that uses g729 when talking to
their SIP provider to minimize bandwidth usage. We are going to want to
be able to record the calls using AMI monitor actions into wav sound
files. All the phones are soft phone running on Windows
simultaneous channels?
Rafael Prado
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: segunda-feira, 22 de março de 2010 2:33
To: Asterisk User MailList
Subject: [asterisk-users] Transcoding
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[7.590966] Zaptel Version: 1.4.12.1
[7.590966] Zaptel Echo
- Katerina Borin katerin.bo...@gmail.com escreveu:
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[
Does anyone know of a utility I can use to transcode a group of files
from G.729 format to something playable on a PC (GSM or WAV).
I know I can convert them individually from the CLI, but I have quite a
lot I need to do.
___
-- Bandwidth and
SOX will do it if you install its G.729 format library.
As far as converting a group of files, that's what scripting is for, i.e.
for FILE in `find . -type f -name '*.g729'`;
do
NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g')
sox [some args] $FILE ... $NFILE ...
done
Thomas Kenyon wrote:
Alex Balashov wrote:
SOX will do it if you install its G.729 format library.
As far as converting a group of files, that's what scripting is for, i.e.
for FILE in `find . -type f -name '*.g729'`;
do
NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g')
sox [some args] $FILE ... $NFILE ...
On Tue, Sep 23, 2008 at 4:44 AM, Alex Balashov
[EMAIL PROTECTED] wrote:
SOX will do it if you install its G.729 format library.
As far as converting a group of files, that's what scripting is for, i.e.
for FILE in `find . -type f -name '*.g729'`;
do
NFILE=$(echo $FILE | sed
I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM
format. I have these same prompts in another server with Asterisk 1.4.18, on
this server the prompts sound pretty nice, but on the first one they sound
pretty choppy. Was there any changes on the transcoding code between this
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk without looking at a change-log, but I highly
doubt a serious modification to the gsm code took place between sub-
versions.
I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM
the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is
pretty bad. The softphone is in the same LAN as the Asterisk server, so I
don't think it's a bandwidth issue.
Best Regards,
On Wed, Aug 6, 2008 at
I am a **BIG, BIG** fan of OpenSUSE.
:)
Use yast under 'Software Management' and do a search for 'gsm'.
Make sure gsmlib and gsmlib-devel are *both* installed. Then scroll
down and make sure that libgsm and libgsm-devel are *both* installed.
After that, you'll have to recompile Asterisk.
Guilherme Loch Waltrick Góes wrote:
I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some
prompts recorded in GSM format. I have these same prompts in another
server with Asterisk 1.4.18, on this server the prompts sound pretty
nice, but on the first one they sound pretty choppy.
On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote:
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk without looking at a change-log, but I highly
doubt a serious
I have used virtually all versions of Asterisk 1.0+ (literally, either
in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel
and haven't had any issues with gcc optimizations with regards to
audio sounding choppy. This scenario for me has always been the gsm
libs.
On Wed, Aug 06, 2008 at 10:15:00AM -0500, Tilghman Lesher wrote:
On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote:
I would make absolutely sure you've got your linux distro's version of
libgsm installed. I can't really speak to the difference between those
two versions of Asterisk
Anyone encountered this on yet?
WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit
frame type 64, while native formats is 4 (read/write = 4/4)
Started after an upgrade from CVS 8/2005 to current 1.2.12.1
If I had a reference for what frame types 4 and 64 are I might
be
Damon Estep wrote:
Anyone encountered this on yet?
WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit frame type
64, while native formats is 4 (read/write = 4/4)
Started after an upgrade from CVS 8/2005 to current 1.2.12.1
If I had a reference for what frame types 4 and
I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below:
Jun 14 09:38:12 WARNING[18292]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/3004-fcfb(256) to SIP/3003-c1c3(2)
Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586
Contact Digium to purchase a G729 license.
Osama Kamal wrote:
I am having a problem with asterisk transcoding GSM and G729 codecs, the
error message is below:
Jun 14 09:38:12 WARNING[18292]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/3004-fcfb(256) to
Hello,
I have an asterisk server running with 23 g.729 licenses. I have
also purchased a sound file from thevoice.digium.com. I need to
covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use
with an IVR system. Is there a way I can convert the files using the
g.729
I don't know about g.729, but this will work for wav - g711.
sox file.wav file.ul
Doug.
-Original Message-
From: Matthew Crocker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 06, 2006 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
- Non-Commercial Discussion
Subject: [Asterisk-Users] Transcoding g.711 - g.729
Hello,
I have an asterisk server running with 23 g.729 licenses. I have
also purchased a sound file from thevoice.digium.com. I need to
covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use
- Matthew Crocker [EMAIL PROTECTED] wrote:
I have an asterisk server running with 23 g.729 licenses. I have
also purchased a sound file from thevoice.digium.com. I need to
covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use
with an IVR system. Is there a way
Kai,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I dont have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
From [cisco (g729)] [asterisk (sip
Hi all,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I
discover I dont have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this
transcoding done.
My set up
From [cisco (g729)]
ADEGOKE ARUNA wrote:
I didn't want to bother the list too much. However, after reading I discover
I don’t have a clear cut way of doing transcoding.
You have posted this three times to the ss7 list and now twice here...
we get it, you don't know how to read the information on the wiki about
Hi all!
Is it possible to have a setup with a server only
dedicated for transcoding from ulaw/alaw to G729. What is the capacity of a
server like that in simultaneous calls?
Regards
Anders
___
--Bandwidth and Colocation
Quoting Anders Svensson [EMAIL PROTECTED]:
this page might help.
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning
Hi all!
Is it possible to have a setup with a server only dedicated for transcoding
from ulaw/alaw to G729. What is the capacity of a server like that in
Why didn't I think of using that command...
It shows all - for G729a which is presumably why I'm having a problem
That would be a problem.
I have purchased 20 licenses from Digium, downloaded binary, registered the
binary correctly,
placed it in the correct directory and it is listed
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite
happy to accept a call from a SIP device using G729a and then complains that it
can't translate into G711 to go onto the ISDN network. Does
AFAIK you need a license from Digium if you want to transcode to/from
G729a...
Hope this information is correct and it helps
Regards
Guido Hecken
I though that Asterisk would transcode between codecs! All my SIP devices
support
G729a 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call from a SIP
device using G729a and then complains that it can't translate into G711 to go
onto the ISDN
network.
Message is no translator path exists for channel type CAPI (native 8) to 256
[EMAIL PROTECTED] 19/07/05 13:44:29
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call
Rich Adamson wrote:
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call from a SIP
device using G729a and then complains that it can't translate into G711 to go
What does your 'show translation' look like?
Can you copy/paste the specific *.conf entries for the sip devices
and capi?
Message is no translator path exists for channel type CAPI (native 8) to 256
[EMAIL PROTECTED] 19/07/05 13:44:29
I though that Asterisk would transcode between
Yes, I've purchased 20 G729a licenses and I know that * uses them OK
[EMAIL PROTECTED] 19/07/05 13:27:04
Rich Adamson wrote:
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to
Why didn't I think of using that command...
It shows all - for G729a which is presumably why I'm having a problem
I have purchased 20 licenses from Digium, downloaded binary, registered the
binary correctly, placed it in the correct directory and it is listed
specifically in SIP.conf
I'm sure
no translator path exists for channel type CAPI (native 8) to 256. I
understood that Asterisk
You don't mention whether you successfully registered the licenses.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Tuesday 19 Jul 2005 14:45, Martin Sutherland wrote:
Why didn't I think of using that command...
It shows all - for G729a which is presumably why I'm having a problem
I have purchased 20 licenses from Digium, downloaded binary, registered the
binary correctly, placed it in the correct
On Tue, 2005-07-19 at 17:05 +0100, Bob Goddard wrote:
On Tuesday 19 Jul 2005 14:45, Martin Sutherland wrote:
Silly question, you did restart * when you put the .so in the correct
directory (normally /usr/lib/asterisk/modules) and it has the correct
permissions?
Does show g729 respond with
I have just purchased 20 licenses for the G729a codec from digium and set about
changing the defaults to use this codec in all cases to reduce the bandwidth
requirements (all my SIP devices support this codec). To my dismay I then find
that calls coming from SIP devices to the outside via the
My VOIP carrier is using G723.1 Codec, so I have set my SIP softphone to G723.1, but I have also set up a Prepaid Calling Card application, which requires a number of sound files to be played. Due to licensing issues sound files on GSMcan not be played because the SIP softphones are on G723.1
Hi, my setup is like:
phones (g729/g711)--(SER)-- Asterisk --(oh323)--gateway (supports
g729g711)
problem begin when phone supports only g711 and Asterisk doesn't
negotiate this codec in full path (from phone to gateway), but tries to
do transcoding (and because I haven't g729 codec in
I apologize if the subject of this message is not the correct one.
My question is: does anyone have any statistics as to how an asterisk
box will behave transcoding SIP calls to IAX2 calls? How many
simultaneous calls can it handle?
Thanks,
Daniel
___
On what trascoding time depends on?
I started server, run * and issued command show translations
--
sipsrv1*CLI show translation
Translation times between formats (in milliseconds)
Source Format (Rows)
Natsvlishvili
Sent: Thursday, 28 April 2005 11:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Transcoding times
On what trascoding time depends on?
I started server, run * and issued command show translations
I am a
little confused aboutvoice data transcodingin Asterisk. I can make a
call between twou-law-only phones over an IAX GSM-codec link and the two
Asterisk servers handle thetranscodingulaw-GSM...GSM-ulawfine.
However, over a SIP channel, this doesn't seem to work. Asterisk appears to be
Before I put myself to the task (next month, maybe) of surveying the
CPU costs of transcoding, perhaps someone else has already done this
work and would be willing to share it or refer me to a link of
previously published data. My reviews of the mailing list with
various keywords were
I have a Budgetone and an ATA but none of them
support GSM. I´d like to place call to the PSTN with my X100P viaa WAN
(64kbps). g711 is out of the question. Can * transcode from g723.1
to GSM? How costly is it? I have tried different configurations on
sip.conf and extensions.conf but have
:40 PM
Subject: [Asterisk-Users] Transcoding
Hello,
Does asterisk do transcoding when the call goes
through the system, codecs are the same but signaling protocol is changed.
example:
SIP with GSM --- IAX with GSM
What quality destruction happen when I use transcoding? I know
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