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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 codec or can I use the one coming...

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker...

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the others. If I disable G.729 on s...

hi all according to what I've found out this far, the G.729 patent seems not valid in a broad range of countries. so... does anyone know where I can find the algorithm? roy

how to make g.729 preferred, but failover to gsm I've purchased a few g.729 licences, and would like to set up iax.conf such that g.729 is used if they are available, but then it fails over to gsm. I'm not sure how to specify such a preference. I'll let the server transcode from ulaw (from the sip p...

Hello I have a confusion relating to G.729 codec. I know how to install where to get license but i really don't know why we need it? Why people use G.729 codec with asterisk? look all functionality can be done with out it ie calling from sip to iax protocol and sip/ iax to E1, then why we need this?? regards Adam -------------- nex...

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> > Message-ID: <479F73A4-7938-4EFD-99B8-EEED7E4ED70C@nosignal.org> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; forma...

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

(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 use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38...

Greetings list, I have an older Dell Poweredge server running Asterisk 1.2.13. I have installed 5 licenses for G.729 from Digium. I have 5 SIP trunks through a US provider. When my system makes outgoing calls, they go out as G.729. However, when an incoming call comes in, my server does not indicate to the provider's server that G.729 is an option, so the remote server sends the call in ULAW. My sip.conf...

...mi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly chea...

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 t...

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 like SJPhone is using a version of G.729 other than Codec G.729 (Annex B) , probably 'Annex A'. What we want is to have a smoother communication when we use a SIP phone on the Dialup conn...

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 through * However if I say canreinvite=no in the sip.conf for either of these gadgets, the call will fail with No compa...

Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist...

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 in, Asterisk does not like the G.729 anymore and will not send the c...

Hello Jean-Marc: I've been watching the speex development from its inception because I and several Tech Startup Connection members have a very important application for this voice encoding/decoding. Further, we are quite familiar with ACELP as implemented in G.729. As far as I know Speex is also ACELP ... yes/no? Question ... what do you see as the advantages of ACELP compared to MP3? I had thought that ACELP yielded much smaller files, but now I'm not sure because I've listened to what I think were 16kbps MP3 files (????) with Apple's ITune...

Upon testing the g.729 licenses in our lab we found the following issues in a plain SIP environment: 1. When g.729 licenses are used in leaving a Voicemail, they do not get released upon clearing the channel. 2. If the voicemail.conf has options for 2 format types (ie. wav49 and WAV), it consumes 2 g.729 licenses. I...

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows "(ULAW)" (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody sugge...

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.p...