similar to: More questions. Call Waiting and Threeway

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'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 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

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:

any idea on how we can setup threeway calling in * thanks moe smadi

I think I found another interesting 'feature' with threeway calling. If you hang up while on a 3 way call with both parties on outside lines, Asterisk ends up removing the conference initiator and leaving the outside trunks bridged together. Is this a good idea? This could cause congestion problems on small configurations with limited outgoing lines. Maybe we should add an option to

I was just wondering if there was a way that you could have two calls on one line and switch between the two without initiating a threeway conversation? I would imagine that Flash is the way to do this, but when I Flash twice, a 3-way call is initiated. If I turn threeway off, then I can't transfer. Also, is it possible to hang up one of the calls, and then continue talking to the second

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the

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 running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the

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

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 I have downloaded a FLAC file somewhere and when trying to decode it to WAV it gives the error message: ERROR, MD5 signature mismatch So my question is now: are FLAC files that give the error message above still decodable to WAV (and how can you do this, because flac.exe doesn't want to decode the file), even if there is a MD5 signature mismatch, or is this not possible at all? thx

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

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

Hi, I have asterisk 1.2.18. I just took a peak at the command: > show translation and I saw that I can only convert from/to ulaw, ulaw, gsm and slin. No speex, no ilbc ... do I need a license or compile something extra? The G723, 726 and 729 ... I need a license, is that it? one for all of them? or for each? How do I get them to work? not just pass-through ... I need conversion. Thanks a

Apologies for the double post in re my SPEEX issues, but can anyone report successfully installing libOgg and getting SPEEX to work on Asterisk? Details: Fedora Core 3, Asterisk 1.0.4, VoicePulse Connect. Here's my original query with the CLI log http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088225.html Thanks. /rg