similar to: Digium G.729 codec now available for Solaris/SPARC

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

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

Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or

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

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

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

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hello, All! We was upgrade our Asterisk from version 0.7.2 to 1.0.7. And have big problem. When asterisk starts: ------------------------------------------------------- *CLI> h.323 show codecs Allowed Codecs: Table: G.729A{sw} <1> G.729{sw} <2> G.723.1{sw} <3> G.711-uLaw-64k{sw} <4> Set: 0: 0: G.729A{sw} <1> G.729{sw}

need transcoding to a|ulaw. I am using it with no problems (have g729 licenses as well though). A bit off topic, I have found some extra configuration that is not really in the docs (or I could not find them): fullname=Your full name country=gr language=en city=City province=Province phone_home=+fullinternationalnumber phone_office=+fullinternationalnumber email=your at email.com

Hi, I am taking over an asterisk system from another person and having an issue where a sip trunk is restricting the outgoing codecs to just g.729 I am dialing in from a Cisco 7960. The Invite from the Cisco has the folowing M line: m=audio 17022 RTP/AVP 18 0 8 101. So it is allowing g.729, ulaw and alaw. Asterisk is tandeming the call out over a SIP trunk sip.conf tandem trunk config:

Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the "old" binary) But there're a few questions: - should not the codec listed in the codec list when i enter "show codecs" ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between

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

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

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

List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all ; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816

On Mon, 2007-02-05 at 12:00 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Mon, 5 Feb 2007 11:36:28 -0500 > From: Andy Davidson <andy@nosignal.org> > Subject: Re: [asterisk-users] Question on G.729 (was: H.264 *Not > Patented*) > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >