similar to: Voicemail Codec challanges.

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'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

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

Ok I have an * server finally setup and acepting calls from freshtel and I am VERY impressed at how well the Freshtel.net service works but thats another subject :) I have it all setup so that I can Dial my DID number on freshtel and that gets set to my * via IAX. At the moment I have the demo configured so that I can test it all and make sure it is all working. The problem is that I

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

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

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

Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg0000.gsm

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

I can't check the voicemail for the switchboard. Asterisk hangs up for some unknown reason... ----- s n i p ----- -- Executing [*500 at default:1] Wait("SIP/597-00f0c410", "1") in new stack -- Executing [*500 at default:2] VMAuthenticate("SIP/597-00f0c410", "500 at default|s") in new stack -- <SIP/597-00f0c410> Playing

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to

Asterisk 1.2, Fedora Core 4: When I leave a voicemail message, it writes the necessary files to the INBOX: msg0000.gsm msg0000.txt msg0000.wav msg0000.WAV When I hang up, the files are erased. There is no indication of anything untoward in the logs: -- x=0, open writing: [...]/INBOX/msg0000 format: wav49, 0x99ce778 -- x=1, open writing: [...]/INBOX/msg0000 format:

An outside caller started to leave voice mail. The CLI shows: Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: gsm, 0x8295d40 -- x=1, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: wav, 0x829e2c0 -- User cancelled by pressing 0 -- Playing 'vm-saveoper' (language 'en') Later on,

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: file.c:1017 ast_streamfile: Unable to open

We've been testing (aim:frziegler and aim:end1r) the Voicemail2 app for a few days now, based on a CVS build from Monday, 5/12/03-23:15. Works good! Thanks Mark! We seem to have found a bug in the MWI (Message Waiting Indicator) logic. By simply creating msg0000.txt files in both structures, e.g.: for extension 4000: voicemail1: /var/spool/asterisk/vm/4000/INBOX/msg0000.txt

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my surprise the messages are indeed

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v