similar to: 1.6.2.6: can't upgrade from 1.6.1.18

Hi Everyone, I am still unable to start Asterisk 1.4 that I installed using ATrpms. I was initially suspecting some permissions issues but it seems to me that its more to do with a speex codec not loading properly. Here is the message I get if I run asterisk -cvvvvvvvvvvvvvv app_userevent.so => (Custom User Event Application) == Parsing '/etc/asterisk/codecs.conf': Found --

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

Hello, I'm using Speex1.1.11.1 source code. I enabled dereverb by speex_preprocess_ctl(). I try different dereverb_level and dereverb_decay values and it seems to not work. I generate some test files in CoolEdit using reverb or echo and then preprocess it by speex_preprocess(preprocess, input, NULL); and output is the same as input:(. Denoising and VAD works very good. I tried dereverb with

Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. -- Executing [s at fax-tx-test:3] SendFAX("SIP/side-sip-00000009", "/var/spool/asterisk/fax/20091113_1455.tif") in new stack [Mar 20 17:05:34] WARNING[6433]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped

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:

Hello I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I don't need: ================ > cat modules.conf noload => codec_speex.c ip04*CLI> reload ip04*CLI> show modules codec_speex.so ================ Just to check, I added the actual filename (.so): ================ > cat modules.conf noload => codec_speex.c noload => codec_speex.so

I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. *************************************

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

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

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

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,

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

Hi all... I have a slight problem with getting speex running I Downloaded Speex sources (v. 1.0.4 stable version) and did make; make install sucessfully Then I re-maked the asterisk sources and clearly saw a speex.so module being built (so the makefile for sure detects that there is a speex lib installed now) After that when I run asterisk: [codec_speex.so] Feb 22 09:32:59 WARNING[29189]:

Hi, Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a way that I cannot script non-english sound files downloading anymore. The following used to work (unattended) with 1.6.1.9 (for instance): cd /usr/src/asterisk-${ASTERISK_VERSION} ./configure make menuselect.makeopts echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >

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

Jean-Marc Valin wrote: >> gcc version 3.4.5 (mingw special) > > Can you try 4.0 or 4.1 just to be sure? mingw (native gcc for win32) doesn't officially support using gcc 4.0 and 4.1 yet (apparantly there have been some issues), so there are no binary packages. But if you think it helps, I can compile gcc 4.1 and give it a shot. >> Compile flags: >> DEFINES +=

I'm having trouble with the preprocessor's noise reduction feature. The basic issue is that it simply doesn't work very well. With my laptop (whose microphone is otherwise quite capable) I routinely hear transient background noise, typing, and other "quiet" sounds leaking through to the speex stream. Even worse, the AGC feature is blowing these things up into just awful

The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by