Displaying 20 results from an estimated 1300 matches similar to: "Speex codec problem (unresolved ?)"
2004 Dec 06
0
Voicemail Codec challanges.
Just working on Configing up Voicemail and now that I have got it
working and configed and answering the way it should be I have another
challange.
on the * CLI> I get this
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/6001/INBOX/msg0000 format: wav49,
0x8133390
-- x=1, open writing:
2005 Sep 07
1
Speex codec - Out of buffer space
Hi,
I'm running Asterisk 1.0.7 and would like to add Speex support. I
downloaded Speex 1.0.5, installed and recompile Asterisk again.
Now trying from X-Lite to connect using Speex but getting lot of weird
erros on Asterisk console:
Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein:
Out of buffer space
I was trying to setup Speex on my second Asterisk server and wanted to
2003 Aug 26
1
More questions. Call Waiting and Threeway
I can't do threeway from my Grandstream phone. Looking through the server
config files, I figured out why - zapata.conf has Threeway turned off for
the channels I use.
I do my work on someone else's Asterisk box and don't want to modify
zapata.conf for several reasons, the biggest being that the guy who owns
the box has a couple clients using it and I am deathly afraid of breaking
2003 Jun 17
11
Speex
Hello everyone.
I am having problems getting speex support.
It seems * is not loading speex. When i did a make in the codecs sub dir,
the following error pops up when making speex:
codec_speex.c:34:19: speex.h: No such file or directory
is this file missing in the cvs as i just removed the whole * dir and did a
new checkout and still seem to get this error, or do i need to get/install
2005 Jan 05
0
Re: Speex codec problem (unresolved ?) = Fixed
>>
>> After looking at the source I had also tried to increase the buffer size
>> from 8000 to 16000, but that made other codecs (like lin_to_g729) choke,
>> and
>> I still had the problem...
>>
>> I like speex and would like to use it (as I find ilbc a bit too scratchy)
>>
>> I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4
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,
2004 Sep 30
0
Oops, a seg fault =(
Ok so it seg faults when I try to dial out through IAX(voiptalk.org),
ofcourse it doesn't if I remove allow=speex :P
----
(gdb) run -c
Starting program: /usr/sbin/asterisk -c
[Thread debugging using libthread_db enabled]
[New Thread 16384 (LWP 28283)]
[New Thread 32769 (LWP 28285)]
[New Thread 16386 (LWP 28286)]
[Thread 16386 (LWP 28286) exited]
[New Thread 32771 (LWP 28287)]
Asterisk
2004 Sep 13
1
problem with dynamic speex library under windows
Hello.
I'm having problems with the dynamic library of libspeex under win32. I
have used the static library for a while with no problems. When I try to
compile my application with the dynamic library I get the following link
error:
codec_speex.obj : error LNK2001: unresolved external symbol _speex_uwb_mode
codec_speex.obj : error LNK2001: unresolved external symbol _speex_wb_mode
2005 Jul 16
2
Memory leak in asterisk CVS
Hi,
My Asterisk CVS is apparently not doing much (other than keeping SIP &
IAX2 registrations alive and doing some ZAP calls (without
echo-cancellation), but slowly the memory is filling up, so much so that
100m virtual memory is used up within 12 hours and I have to restart the
asterisk application every 48 hours to make sure I have enough memory...
How can I help resolve this problem?
2004 Jul 13
1
bad sound quality, also the ringtone
Hi,
it took me 2 days to get my asterisk box running, so now I completed and
I am disappointed of the sound quality. When I call other people their
voices sound somewhat scratchy. First I thought it might be a codec
problem, but I also recognized it during the ring tone or even the DISA
connect tone. Sometimes it is better quality and sometimes more scratchy.
Where might be the problem? I am
2009 Jan 07
2
\iaxclient-2.0.2 compile problem
Hi,
I had downlaoded iaxclient-2.0.2 and complie project
*\iaxclient-2.0.2\contrib\win\vs2005*
**
It gives many83 fatal and file missing error of file missing
Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such
file or
directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c
40
Error 2 fatal error C1083: Cannot open
2010 Nov 07
2
"scratchy" sound on TE410P
asterisk 1.4.35
dahdi 2.3.0.1+2.3.0
one span on a 4port T1 card
Got complaints this morning that outbound and inbound calls were
"scratchy" and I made a few test calls. It kind of sounds like the gain
is too high somewhere, and the audio is overdriven. Is this a problem at
the carrier? I'm trying to call them now, but it's Sunday morning in the
sticks, and my chances of
2004 Oct 17
2
Anyone else tried Speex 1.1 CVS?
I built the CVS version of the Speex library - v1.2 it calls itself.
Asterisk seg faults trying to use codec_speex.so.
I'll have a look to try to fix it, but thought I'd just ask if anyone else
knows what needs to be done?
Steve
2004 Jul 29
6
Zaptel doesn't see remote hangup ? euro-isdn
Hi
Just received my spanky new TE405P today to replace my Cisco gateway...
After much fiddling (I forgot to switch it to E1) I got it to work and
everything "seems" to work perfectly on our ISDN PRI.
If I dial-in from the PSTN to a SIP phone, the call goes through and if I
hangup either the SIP phone or the remote end, the call gets disconnected
and destroyed
However, if I dial-in
2010 Apr 09
3
scratchy sound
Hi,
I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens.
Please listen to the following sound file:
http://213.96.91.201/temp/distorted_audio_1.wav
This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to
2010 May 13
2
LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
Hi,
I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other IAX2.
Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of
2004 Dec 05
2
ANALOG FXO ZAPTEL & WCFXO & WCTDM module issues seen with intermittent analog lines
Hello, I have found a "bug", I think in the way TDM400P cards handle FXO
interface disconnect/re-connect problems. Normally I do keep all the wires
connected from my CO / PABX quite securely, but I had a need to re-route the
cable from one side of the desk to another, and I simply disconnected the
RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY
SCRATCHY AUDIO
2003 Aug 19
1
Speex & openh323
hi,
I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with
2004 Jul 15
3
G.729 codec doesn't seem to work *even* after installing the license
Hi,
I am trying to post this again as I am getting no answers and the
support@digium.com bounces...
(I have searched the whole list and can't find the answer either)
I have installed a 5 user license for G.729 and want to route calls through
Asterisk from my G.729 phone to Cisco AS5300 also using G729.
Both Cisco and the phone connect using this codec if I do not force the call
to go
2004 May 14
1
Dead FXO Module on TDM400P?
Since the irc channel wasn't any help, I will try posting my problem here.
I have two TDM400Ps less than a week old in a PC. All of the FXS ports
work fine, and all of the FXO ports worked fine up until thisafternoon. If
I try to dial in, I get a busy signal, if I try to dial out, all I hear
is a very scratchy, very crackly dialtone. If I swap the first FXO module
with the second on the card,