similar to: Problems with Licensed g729a codec from Digium

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Hi, I have a Cisco 5300 which I want to make a call THROUGH the Asterisk PBX (security) to an IP phone which supports g729, and vice versa. Both Cisco and the phone talk this codec if I do not force the call to go through * However if I say canreinvite=no in the sip.conf for either of these gadgets, the call will fail with No compatible codecs! I have bought a 5 user license just to

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

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

List, I purchased 2 g729 licenses but I can't get it to answer a g729 call from a cisco router with a vwic card. In the debug output below you will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263) when it should support g729 according to the config also listed below. The real odd thing is I can place g729 calls to the router, just not from the router to *. Anyone have any

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

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

-----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

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, 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

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, 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

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

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

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

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

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23

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