Displaying 20 results from an estimated 11000 matches similar to: "call transfer problem - something strange"
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
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
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
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
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]:
2007 Jun 27
0
Asterisk to Cisco 2600 GW DTMF Not Working, Working now
On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote:
> 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
2004 Jul 08
0
Problem SIP no audio just noise
I'm trying to call from XLite phone to PSTN
(I've tried this from internet and from local network the same)
The Xlite doesn't write that it is connected but receives excelent audio.
At the other end comes only noise. Some times only for a second you can
here the
caller voice , but this was only one time :)
I saw with ethereal that UDP packets are coming and going to the
asterisk
2006 Feb 13
1
iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)
Hello all,
I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of them:
codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long
from RTP (38)?
The ATAs in question are various Grandstream models - the HT486 being the
predominant one. Looking at the list archives, it's
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
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,
2008 Apr 02
0
Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released
The Asterisk development team has released version 1.4.19 of Asterisk and
1.6.0-beta3 of Asterisk-addons.
The new Asterisk-addons release contains a few bug fixes over the previous version.
http://svn.digium.com/view/asterisk-addons/tags/1.6.0-beta3/ChangeLog?view=markup
Asterisk 1.4.19 contains a large number of fixes over the previous release,
1.4.18. For a full list of changes, see the
2008 Apr 02
0
Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released
The Asterisk development team has released version 1.4.19 of Asterisk and
1.6.0-beta3 of Asterisk-addons.
The new Asterisk-addons release contains a few bug fixes over the previous version.
http://svn.digium.com/view/asterisk-addons/tags/1.6.0-beta3/ChangeLog?view=markup
Asterisk 1.4.19 contains a large number of fixes over the previous release,
1.4.18. For a full list of changes, see the
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
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 Mar 20
1
I cannot use G711 (ulaw|alaw)
Dear all,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to,
I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect
attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with
our capability 0xfe02.
I do not understand why because my Asterisk box load these codecs properly!
Does somebody
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
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