similar to: Asterisk Not Starting after YUM Update

I'm trying to use the Digium suplied G.729 Codec, I have ran the register utility, and got my licenses written to /var/lib/asterisk/licenses, but when a start Asterisk I got the following errors: [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: This module is supplied under a

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File

Hi, I carefully followed instructions in README file lasting with : /root/register ... blabla asterisk -r CLI> restart now Then asterisk -r fails with : # asterisk -r Asterisk 1.6.1-beta4, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is

Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of

Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual

Hello, I wonder if anyone else has noticed this. I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have a leaked file descriptor that remains until asterisk dies. Now, maybe no-one sees this, mainly because I have no g729 licenses on the machines where this happens. And conversely, I haven't yet studied servers that do have licenses. Why have codec_g729a.so loaded if

Hi, I've been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. I've removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a

Recently with Asterisk 1.4.17 I've been running into some stability issues. I started looking through my logs, and I found this: [Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module 'codec_g729a.so' was not compiled against a recent version of Asterisk and may cause instability. I'm using the newest version of codec_g729a.so from the Digium website (v33).

Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in

Hello! It seems that the new codec is not backward-compatible to glibc 2.1/2.2 so I receive the following error: [codec_g729a.so]Oct 22 02:37:18 WARNING[3433]: loader.c:325 __load_resource: /u sr/local/glibc/libc.so.6: version `GLIBC_2.3' not found (required by /opt/files/ usr/lib/asterisk/modules/codec_g729a.so) Oct 22 02:37:18 WARNING[3433]: loader.c:554 load_modules: Loading module

Hello Everyone, I just upgraded to the latest beta version and I am running into one problem. We purchased g729a licenses from digium and they aren't loading anymore. If I roll back asterisk to 1.2.10 the codecs work fine. I've downloaded the new 1.4 version of the codec from their website and re-registerd everything with no luck. Here is the error message: error loading module

Hi all, I have install G729 license to asterisk 1.4.18 and use distro debian etch 4r2. It'snot complete. when i use ldd commant to show the library it show "/usr/lib/aserisk/modules/codec_g729a.so: /lib/libc.so.6: version 'GLIBC_2.4' not found (required by /usr/lib/asterisk/modules/codec_g729a.so) when i check the GLIBC version on debian it is glibc-2.3.6 please advice me

Hi all, sorry bothering again. I am still stuck in compiling asterisk. Learning (or trying to) from the first problem and first hint, when I got this error: gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out

asterisk-users-request@lists.digium.com is believed to have said: >Hi, > >Sorry for being very late on this thread but i am trying to make a >decision on which one to go for. Options are > >1. Dock n Talk offered by Voxilla (USD139) >2. GSM Gateway by CyberTelecom (GBP60) > >I'm having a TDM400P with 1 FXO & FXS. > >I'm interested in implementing DISA

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

asterisk-users-request@lists.digium.com is believed to have said: >I don't mean to be rude to everyone who responded to this question, but >I think that everyone is answering the wrong question. The point is that >the message waiting indicator doesn't light up, at all, ever. All that >happens when messages are waiting is that the display blinks and the >phone gives a

Hi all, I have noticed something I can't solve regarding Asterisk (latest 1.6.0.x). My server is set at the GMT+2 timezone. The clock is ok (I can get the correct time at the terminal). But today I got a call at a time where Asterisk should have gone 'off business hours'. All log times are wrong by exactly 2 hours. As if Asterisk would just sit on GMT, ignoring the GMT+2