Displaying 20 results from an estimated 5000 matches similar to: "iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)"
2008 Apr 13
1
compilation of asterisk 1.4.19 with ilbc already on system
I already have ilbc installed on my system. The files
are:
/usr/include/ilbc/iLBC_decode.h
/usr/include/ilbc/iLBC_define.h
/usr/include/ilbc/iLBC_encode.h
/usr/lib/libilbc.a
/usr/lib/libilbc.la
/usr/lib/libilbc.so -> libilbc.so.0.0.0
/usr/lib/libilbc.so.0 -> libilbc.so.0.0.0
/usr/lib/libilbc.so.0.0.0
However, if I do a "make" in asterisk-1.4.19, it will
not detect that libilbc.a
2004 Sep 25
1
ilbc problem
Hello,
I'm going to use * as SIP<->H.323 proxy (codecs doesn't matter - only
pass through). I compile * (v1.0.0) without any problems as far as
H.323 stack (pwlib, etc). But when I'm trying execute asterisk -vvv I'm
getting error message:
[codec_ilbc.so]Sep 25 15:15:43 WARNING[16384]: loader.c:248
ast_load_resource: /usr/lib/asterisk/modules/codec_ilbc.so: undefined
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
2007 Mar 14
1
strange things on call transfer
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I'm setting up an Asterisk system which is connected to an Alcatel 4400
PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
call by hitting the # key, I get this messages and nothing happens on
the phone:
WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame
that isn't a multiple of 50 bytes long from
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
2011 Sep 19
0
iLBC support in Asterisk after Google's acquisition of GIPS
Recently, we were notified that the mechanism included in our Asterisk
source code releases to download and build support for the iLBC codec
had stopped working correctly; a little investigation revealed that this
occurred because of some changes on the ilbcfreeware.org website. These
changes occurred as result of Google's acquisition of GIPS, who produced
(and provided licenses for) the iLBC
2011 Sep 19
0
iLBC support in Asterisk after Google's acquisition of GIPS
Recently, we were notified that the mechanism included in our Asterisk
source code releases to download and build support for the iLBC codec
had stopped working correctly; a little investigation revealed that this
occurred because of some changes on the ilbcfreeware.org website. These
changes occurred as result of Google's acquisition of GIPS, who produced
(and provided licenses for) the iLBC
2004 Sep 27
0
Speex/ILBC buggy with * 1.0 and X-Lite/Pro?
I'm playing with codecs at the moment and have found some notices errors
when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was
getting garbled sound, but after changing magic number for both codecs
to 97 (as per
http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and
http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to
get normal voice. BUT,
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 Nov 29
2
Cannot Start Asterisk
Hi,
I'm running asterisk-1.0.2-2mdk. When I tried to start it with
/usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvgc, I get
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
Ouch ... error while writing audio data: : Broken pipe
# ps aux | grep mpg123
root 5237 0.1 0.4 5816 4444 pts/0 S 18:45 0:00 mpg123 -q
-s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3
2003 Jun 08
1
anyone seen this error when running asterisk!
Hi all -
I'm making gradual progress implementing asterisk on my box! Now, when I
type asterisk it dies at this point. Does anyone have any idea why this is
happening! It have checked everything but running out of options!
[app_voicemail2.so] => (Comedian Mail (Voicemail System))
== Parsing '/etc/asterisk/voicemail.conf': Found
== Registered application 'VoiceMail2'
2004 Jun 02
1
DTMF and SIP
Hi
I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the
phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also
tried inband) and I get the following error:
june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
This means that I cannot get access to voicemail from the handsets
2005 Oct 05
0
call transfer problem - something strange
Hi,
I try to set up planet VIP-050 with asterisk. Everything works fine
instead of the call transfer. When I press # console says something
like this:
>Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh,
format changed >to 1024
>Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein:
Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP
2007 Jun 26
1
Asterisk to Cisco 2600 GW DTMF Not Working
Hi All,
I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
with a PRI card in it, handing off to a PBX and vise verse. Calls in
and out are working fine except for DTMF from Asterisk to the 2600.
DTMF from the 2600 to Asterisk is fine.
Here are the Asterisk console warnings I get when I send DTMF from
Asterisk to the 2600:
== Forcing Marker bit, because SSRC has changed
Jun
2004 Jul 09
1
RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000
To me it's a error if I can't complete calls using the ATA configured to use
the g726 codec.
I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I
received NOTICES and WARNINGS, but I can't complete a call.
On a zap channel:
-- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack
-- Called 1/2217008
-- Zap/1-1 answered
2006 Feb 20
1
g729 quality at GSM bitrates
Greetings all,
I'm trying to improve the codec selection on a few of the asterisk boxes we
have to keep the g729 licences free for calls from ATAs that don't support
anything apart from g711 and g729. GSM seems to offer noticably inferior
call quality (at least when using a softphone + decent headphones), but it's
about where I want the bitrate to be.
I know there are lots of Speex
2004 Jul 09
3
ATA 186, firmware SIP 3.1 and codec g.726
I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to
use the g.726 codec I received many erros and the calls doesn't work.
I changed the fields:
- LBRCodec: 6 <- the code for g.726
- TXCodec: 6
- RxCodec: 6
The errors:
Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to
calculate samples for format G726
Jul 9 13:15:37 NOTICE[1192491824]:
2006 Oct 23
2
T.38 faxing with spandsp and Grandstream HT.486
Hello !
I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal.
As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine.
Has anybody success with the HT486 as T.38 terminal ?
ATA as originator: I managed only onetimes a successfull T.38 fax
session. The other times the HT486 did not initiate a re-invite with
T.38 parameters. Or shall the Terminator
2004 Jul 13
1
segmentation fault on asterisk startup
Hi,
I write to this list, because I didn't find anything on the net.
I installed asterisk using bristuff-0.0.2 without any errors, but when I
start asterisk with "asterisk -vvvc" I get following error:
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
== Registered translator 'ilbctolin' from format ILBC to SLINR, cost 127
Segmentation fault
Removing
2009 Apr 14
3
Changing menuselect values from CLI and not TUI
Hi All,
I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually.
Is there a way to make the changes to menuselect from the CLI?
As an example, selecting the iLBC codec.
menuselect codec ilbc on
Regards
David.
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