similar to: FreeBSD Digium g.729 codec seg faults on rev 30652

Trying to update to dovecot-1.0.alpha5 and seeing this at compile time: mech-gssapi.o mech-gssapi.c; then mv -f ".deps/mech-gssapi.Tpo" ".deps/mech-gssapi.Po"; else rm -f ".deps/mech-gssapi.Tpo"; exit 1; fi mech-gssapi.c:30:27: gssapi/gssapi.h: No such file or directory mech-gssapi.c:42: error: syntax error before "gss_ctx_id_t" mech-gssapi.c:51: error:

As of test-60 and test-61, running Squirrelmail 1.5.1[cvs] on FreeBSD 5.3-STABLE, messages are not marked as read after they are read. If I switch back to test-59, it works normally. When switching back to test-59, with log_path = /var/log/dovelog dovelog shows: dovecot: Jan 14 11:43:38 Error: IMAP(kimc): Corrupted index cache file /home/kimc/Maildir/dovecot.index.cache: indexid changed

Hello , I'm have this problem before copy codec in the /usr/lib/asterisk/modules before registration ... My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running Linux on 2006-04-20 01:02:07 UTC This erro : codec_g729a.so]May 5 21:39:16 WARNING [6950]: loader.c:731 __load_resource: misstng mod_data for codec_g729a.so Segmentation fault Thanks

Attempting to setup smtpd auth for Postfix using the Dovecot authorization unix socket. Some info is at: http://www.postfix.org/SASL_README.html To enable operation when Postfix is chrooted, a method from the reference above is to locate the socket at: /var/spool/postfix/private/auth To do this, I have this configuration in dovecot.conf: # listener sockets will be created by Dovecot's

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

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

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

Firstly, I'm running Asterisk 1.4.4 on Solaris 10. I have several different internal SIP phones all sharing a single IAX2 VoIP channel. PHONES |------------- <SIP/uLAW> --------------| ASTERISK |-------------- <IAX2/g729> ------------|VoIP/ISP The g729 codec has been registered successfully and appears to be detected by Asterisk (NOTE: I have changed what I thought might have

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

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > 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

I have been shaking down a dialplan for SIP fax to efax. The basic senario is an ATA on the same subnet as the Asterisk 1.2 box (avoid RTP packet lose and thus fax crash), calling a 'fax extension' and envoking rxfax then email. I leverage off of context: from-internal-additional-custom, so as not to have it overwritten by FreePBX. In extension_custom.conf I have:

Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). ***** A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium,

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

Hello, Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are instructions to install G729 software. (I think I followed instructions step by step but g729 license doesn't seem to show up). My question is : Is the command bellow still up to date ? >g729 show Regards PS: Here are latest steps: # ./benchg729-1.0.6 Recommended flavor for this system is

Somebody can say me what i can do that the g729 is working? _____ Von: Ren? Enskat [Teamware GmbH] [mailto:ren@teamware-gmbh.de] Gesendet: Montag, 10. April 2006 10:21 An: 'asterisk-users@lists.digium.com' Betreff: G729a error when i load asterisk i got this error and cant start * with the g729 codec: Apr 10 10:21:18 VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10:21:18

2009/5/8 Olivier <oza-4h07 at myamail.com> > Hello, > > Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are > instructions to install G729 software. > (I think I followed instructions step by step but g729 license doesn't seem > to show up). > > My question is : > Is the command bellow still up to date ? > > >g729 show I suddenly

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 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

I have the same problem on some modules. For example app_math.so [app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728 __load_resource: missing mod_data for app_math.so Any help?. I have been looking , but nothing reasonable found. Thanks -- Alberto Sagredo

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).