similar to: Anyone else tried Speex 1.1 CVS?

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

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

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

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

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

Hi, I'm sorry to bring this up again, but I have been googling forever and whatever solutions are offered don't work for me. I am using x-lite (the latest build) and trying to use Speex. When I do call from the x-lite to another SIP phone or PSTN (through Cisco gateway) My asterisk fills up with this message: WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space The

The short: Linking to libspeex.a when generating a .so using libtool results in a non-portability warning. This is due to PIC code and non-PIC code intermingling. How can I go about fixing this whilst still using an installed libspeex present on the user's system? The long: I am using autoconf + libtool to generate a codec plugin for speex (sipXmediaLib), and I'm trying to eliminate

http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha1.tar.gz http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha1.tar.gz.sig This is the only v1.2.alpha announcement I'm going to send to dovecot-news list. Next announcement will be for either v1.2.beta1 or v1.2.rc1. I'm hoping that v1.2 stabilizes pretty soon. There are no huge changes like there were in 0.99->1.0 or

http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha1.tar.gz http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha1.tar.gz.sig This is the only v1.2.alpha announcement I'm going to send to dovecot-news list. Next announcement will be for either v1.2.beta1 or v1.2.rc1. I'm hoping that v1.2 stabilizes pretty soon. There are no huge changes like there were in 0.99->1.0 or

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:

Now supported by v2.0. Also as a patch to v1.2: http://dovecot.org/patches/1.2/zlib-compress.diff I'm not really sure if I should commit it to v1.2 code tree. The code contains ugly copy&pasted io_stream_copy() and v1.2 is supposed to be feature complete.. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature

http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha2.tar.gz http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha2.tar.gz.sig Mainly the same fixes as in v1.1.3 -> v1.1.4 plus some more v1.2-specific fixes. Not many people have been testing v1.2 releases. Wonder if there's something I could do to improve that. I guess v1.2 is kind of missing those killer features that would get a

Hi, Has anybody some information on on the "Redundant audio data" header in the speex rtp payload? Is this header always present? Is its value always the same? Can it be modified through some speex_*_ctl function? Thanks, Emmanuel -- ------------------------------------------------------------- Emmanuel Wauters Tel : (+32) 11 30 13 30

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

I am currently running v1.1.20 and was wondering if there is any compelling reason to upgrade to v1.2 or v2.0? For example is there any security updates in v1.2 or v2.0 that aren't included in v1.1.20? The other major thing I would be interested in knowing is there any speed enhancements in the newer versions? I am using basic PAM Auth and Maildir mailboxes on a Linux box. Let me know

I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure

http://dovecot.org/releases/1.1/dovecot-1.1.5.tar.gz http://dovecot.org/releases/1.1/dovecot-1.1.5.tar.gz.sig http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha3.tar.gz http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha3.tar.gz.sig A combined release announcement for a change. :) Largest changes since v1.1.4: * Dovecot prints an informational message about authentication problems

http://dovecot.org/releases/1.1/dovecot-1.1.5.tar.gz http://dovecot.org/releases/1.1/dovecot-1.1.5.tar.gz.sig http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha3.tar.gz http://dovecot.org/releases/1.2/alpha/dovecot-1.2.alpha3.tar.gz.sig A combined release announcement for a change. :) Largest changes since v1.1.4: * Dovecot prints an informational message about authentication problems

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

Hi, Right now, I'm thinking of adding a speex_lib_ctl() call that would support SPEEX_GET_VERSION (and return a string) or SPEEX_GET_MAJOR_VERSION and SPEEX_GET_MINOR_VERSION (and return ints). I'm open to other suggestions though. If there's anything you'd like to see in the API for 1.2, say it now. ...and no, I won't add a speex_do_all_the_work_for_me() call :)