Displaying 20 results from an estimated 1000 matches similar to: "Unable to find a path from G729A to ALAW, Unable to find a path from GSM to G729A"
2006 Jan 14
0
codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
Hi guys,
Anyone seen something like below(see below the line)?
Machine P2 w/512MB RAM
Debian (testing) ; kernel 2.6.12-1-386
asterisk 1.2.1-n-all incl. astcc
For many months now I went through * 1.07, 1.09 and never
saw something like that. Even with 1.2.0, a month now,
at the beginning everything was fine, and suddenly
"codec_gsm.c:194 gsmtolin_framein: Invalid GSM data" thing
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2010 Jul 03
0
[asterisk-user] gsmtolin_framein: Invalid GSM data
Hi
I have created meetme with 3 user. When i going to mute user it gives
following error..
*Asterisk Version : 1.6.2.6*
-- <SIP/52987-00000040> Playing 'conf-muted.gsm' (language 'en')
[Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid
GSM data (1)
[Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not
update samples 0
[Jul
2006 Jan 04
2
suddenly iax calls don't work anymore
Hi,
Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore
with my voip provider. I am not aware that I changed anything in the configuration, but
who knows. Can somebody explain me what is happening here? I changed username,
password and number.
-- Executing Dial("Zap/2-1",
2009 Aug 07
0
asterisk crashes!!!
Hi,
I got ast. 1.6.0.10 working for a few weeks without a problem.
A few mins ago..I got the following msgs on ast-cli and asterisk service
crashed.
I coudlnt find anything that might cause this problem.
Any ideas??
[Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM data (1)
[Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did
not update samples 0
2007 Feb 24
0
1.4.0 spews garbage on CLI, crashes
Hi, I just installed asterisk 1.4.0 on my mac. I compiled from source
with no issues. I installed the sample config files, and basically
just added a register line to sip.conf (to register with a Free World
Dialup account).
Then I called my asterisk system from a different computer (using
x-lite softphone on windows xp, registered to an ekiga.net account).
Asterisk answers, and I can hear the
2004 Jul 13
1
G729A and GSM - newbie question
Hello,
When I'm trying to play standard sound files from
Asterisk using G729A codec with OH323 channel
I get this message:
channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A
It seems that this files must be in G729 format?
How can I convert this files to G729?
... or am I wrong?
--
wbr, Oleg
2004 Dec 18
0
what the heck? codec_gsm.c:135 gsmtolin_framein: Huh?
I park a call and instead of the parked extension
being returned, I get silence and the log shows
a bunch of the following messages
WARNING[26220]: codec_gsm.c:135 gsmtolin_framein: Huh?
A GSM frame that isn't a multiple of 33 or 65 bytes long from
(null) (320)?
what does this mean?
BTW these messages are intermittant. sometimes it works fine
other times i get the above message
Regards
2003 Nov 18
3
"Unable to find path from G729A to ULAW" on Sipphone.com
I seem to be having a problem with transcoding and/or agreeing on a
valid codec. I am running a new image pulled from CVS at 1:30 PM CST.
The issue occurs when I try to make a call to a toll-free number over
sipphone.com.
Here's what I see in the console:
NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1259545280]: File
2008 Nov 13
0
Problems with Licensed g729a codec from Digium
Firstly, I'm running Asterisk 1.4.4 on Solaris 10.
I have several different internal SIP phones all sharing a single IAX2
VoIP channel.
PHONES |------------- <SIP/uLAW> --------------| ASTERISK
|-------------- <IAX2/g729> ------------|VoIP/ISP
The g729 codec has been registered successfully and appears to be
detected by Asterisk
(NOTE: I have changed what I thought might have
2006 Jun 27
1
Help Asterisk crashes
I am getting thousand of these messages in asterisk console
Jun 27 12:35:55 WARNING[16496]: codec_gsm.c:194 gsmtolin_framein:
Invalid GSM data
And after some time the system crashes. Does anyone know why?
I running Asterisk SVN-trunk-r7522 built
Does it help to upgrade the system?
Regards,
Fredrik Jensen
2004 Jan 05
0
Codec Negotiation Does not seem to work as e xpected ?? Help Please !!
Steve,
My Problem is not a problem, with the codec negotiation between end points.
But when asterisk does it with canreinvite=no, * do not do it right. I
replied with a lengthy discussion about my findings here, This behavior can
be reproduced. But '*' do not seem to do the negotiation correctly.
http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html
2004 Jul 29
0
G.729 between Zap and SIP
Hi,
I have licensed the digium G.729A codec. But for some reason incoming and
outgoing calls will ALWAYS use G.711a. When I force my phone to only accept
G.729 then an incoming call from ZAP goes straight to my voicemailbox as the
phone doesn't accept the codec Asterisk wants, even if I force it in
sip.conf.
Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ?
The
2004 Jul 30
0
G.729 <-> ZAP ?
Hi,
I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card.
Incoming calls and outgoing calls between my cisco and my SIP phone works
fine on G.729. Recording messages in the asterisk voice-mailbox also works
fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have
licensed the digium G.729A codec.
When I connect my ISDN PRI to my Zap card and I call
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 Mar 16
1
Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
Hi everyone,
I have an issue which is kind of a catch 22 situation. I had outgoing
calls to my new PSTN provider working perfectly. Then I started
focussing on incoming calls. It seems that I can solve an error which
gets my incoming calls working but that in turns means my outgoing calls
don't work. - Strange.
Anyhow I was getting an error:
Process_sdp: No compatible codecs!
And from
2004 Sep 08
1
Problem playing file with G729A
Hi,
I tried to play the standard demo-echotest file !.
It works when i use an ip-phone (like x-lite or kphone), but as far as i
use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the
following error:
Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format:
Unable to find a path from GSM to G729A
Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:
2004 Jun 07
2
AGI + g729A
Hello....
I have the follow situatuion:
< ISDN >
|
|
V
E100P
|----------------| IAX2 / g729A |----------------| T100P
| Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - -
-> |--------------|
| | | | | Zhone |
----------------- ----------------- ---------------
Here's the situation: I receive calls from the PSTN
2010 Dec 27
1
G729a and G729 interoperability
Hello!
I am wondering how the differences between G729, G729a, and G729b
effect call bridging and server interoperability. For example, can
one server use the G729 code with another server that uses the G729A
codec?
Also, which version is Asterisk set up to use?
Thanks!
Elliot
2006 Mar 13
1
G729A
Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent
sessions that P4 server board that can stand? Pls advise.
Btw, if G729A has been purchased and installed, what will happen to the
Asterisk Server crash say hard-disk when down or faulty, any where to do
back up first such as "tar" commands?
Any advice will be appreciated
tq