similar to: Digium G.729 codec modules updated

This evening I posted a new set of Digium G.729 codec modules to our FTP server and web site, for Linux x86 and x86-64 processors. They were built using GCC 4.0.1, and they now report the processor they were optimized for when they are loaded. The previous x86-64 module required a non-standard Asterisk binary configuration, so this was corrected. In addition, there was only a generic version

The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk). Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues. -- Jason Parker Digium

The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk). Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues. -- Jason Parker Digium

As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous versions. In addition, there are now versions for both Asterisk 1.2 (and previous releases) and the soon-to-be-released

As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous versions. In addition, there are now versions for both Asterisk 1.2 (and previous releases) and the soon-to-be-released

The Digium G.729 binary codec modules (both 32-bit and 64-bit), and register tool have been updated for use with the Asterisk 1.4 beta on Solaris. As always, they can be downloaded from http://ftp.digium.com/pub/telephony/asterisk/g729/ -- Jason Parker Digium

The Digium G.729 binary codec modules (both 32-bit and 64-bit), and register tool have been updated for use with the Asterisk 1.4 beta on Solaris. As always, they can be downloaded from http://ftp.digium.com/pub/telephony/asterisk/g729/ -- Jason Parker Digium

Greetings- Was running the Digium FreeBSD g.729 codec until recently when the latest Asterisk bits were obtained via svn: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk This produced: Checked out revision 30652 This on FreeBSD 6.1-RELEASE Attempting to start asterisk it returns: == Registered custom function URIENCODE [codec_g729a.so]May 27 13:29:59 WARNING[71884]:

I have just uploaded Solaris/SPARC G.729 codecs and register tools to our FTP site; they are located with the other non-Linux codecs in the 'unsupported' directory of the g729 subdirectory. There are versions for both Asterisk 1.2 and Asterisk 1.4 in both 32-bit and 64-bit flavors; we will be updating the other 'unsupported' codecs and also producing Solaris/X86 versions some time

I have just uploaded Solaris/SPARC G.729 codecs and register tools to our FTP site; they are located with the other non-Linux codecs in the 'unsupported' directory of the g729 subdirectory. There are versions for both Asterisk 1.2 and Asterisk 1.4 in both 32-bit and 64-bit flavors; we will be updating the other 'unsupported' codecs and also producing Solaris/X86 versions some time

Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in

I have problems building R 2.11.1 with an external BLAS. I've tried several with several libraries: # ACML: export LD_LIBRARY_PATH=/site/VERSIONS/acml-3.6.0/gfortran64_int64/lib BLAS=--with-blas="-L/site/VERSIONS/acml-3.6.0/gfortran64_int64/lib -lacml" LAPACK=--with-lapack # MKL 11: BLAS=--with-blas="-L/site/VERSIONS/intel-11.1/mkl/lib/em64t -lmkl_gf_lp64 -lmkl_sequential

Asterisk G.729 -------------- Digium is pleased to announce that we now have the G.729 Codec available. This industry standard allows you to stuff more calls in a limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (of course normal IP overhead adds to this number). Many people are using Asterisk

(Note: resending with proper Subject) If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then

Dear all, I have Trixbox 2.6 (Asterisk 1.4) installed in my voip server. I have the following short questions about the usage of G.729 codec: 1) Does Asterisk have installed the G.729 codec by default ??? 2) If I don't want to pay for a codec license, using Asterisk in "pass-through" mode for G.729 voice communications, do I just have to download the open source version of the G.729

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script

Hello, I have trouble activating the G.729 codec in Solaris. It seems the module doesn't try to read the license file at all. Although the module loads (it's listed in show modules), "show g729" gives "No such command" error and this is posted in the messages log: Oct 27 18:09:40 NOTICE[19336] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007

Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I

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