similar to: pass through mode

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51 > Send asterisk-users mailing list submissions to > asterisk-users at

Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order

Dear i have buy two license of G729 codec and i have install/registered as documented but after i start "Asterisk -vvvcng" i notice this warning and if i made call the CLI say "No compatible codec!" How can i solve this problem? Thanks in advance Dimitri ------------------------------------------ [app_datetime.so] => (Date and Time) == Registered application

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

>My CPU is Centaur VIA Nehemiah with 998.715 MHz processor not INTEL CPU. Common Sense dude - The Intel IPPs work only for Intel CPUs. Remove the codec_g729.so file from /usr/lib/asterisk/modules folder and restart asterisk Check if it is working by starting as under: UnixPrompt# asterisk -cvvvvvvvvvvvvvvvvvvf Once this loads correctly use 'Stop now' and restart with command

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

The G.729.1 "wideband" codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it

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> >

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 -------------- next part -------------- An HTML attachment was

Hi, all, I have buy 5 Digium's G.729A codec(it just support G.729A license) When I calll with 2 SIP UA that support G.729A and G.729B, its rtp frame have some problem when softswitch with Asterisk. The voice frame have been drop, so sometime I can't hear voice. If I want to fix the problem when softswitch G.729A and G.729B codec. What source code I must to modify ? Or some people have

Hi, some questions/comments about performance/utility of * and * hardware I've been reading this list for a few weeks and I think I have compiled the better feelings of the users. please correct me if I'm wrong, still learning * .... Will be nice to see something like this in a wiki. After being flamed and corrected I will repost "clean" data. 1- Transcoding is the process of

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

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it

hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run "core show codecs" in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i should buy its license in order to use it. is it true? if

Hi i have buy two license of G729 codec, but after run Registration program i notice this error Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient which contains your machine signature and

The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/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

Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

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