similar to: Software patents (was G723 on asterisk 1.4.1)

Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

AFAIK g729 patent is expiring sometime in 2019-2020. Mitul Limbani On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigoseth at gmail.com> wrote: > Hi Steve, > > I understand your question and your point, but I use the g729 codec from > the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 > without a single problem. > > So, sory but I

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

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

Is anyone selling G729 License elsewhere other than Digium? Anyone allowed to sell a similar License as a reseller? -Kannaiyan

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

Thank you all for your answers. They were all useful. Il 21/11/2011 07:37, Erik de Castro Lopo ha scritto: > Giulio Paci wrote: > >> thank you for your answer. So the problem would be suboptimal >> compression due to suboptimal assumption about the input signal, right? >> What I do not understand is how the format of a FLAC format would be >> affected by supporting

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Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _________________________________________________________________ Use MSN Messenger to send

Dear Jean, The source code for G.729 can be download from ITU for free. Also, some developer can do yourself as open source G.729 codec without any help. In this case each who use this codec which source code is free and open source must pay, but not to the developer. Best Regards, Miroslav Nachev JMV> Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit : >>

Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Most Fax machines do work but I have problems with people having Tobit FaxWare and Shamrock CapiFax. http://www.tobit.com/login/mrd.asp?CategoryID=120 http://www.shamrock.de/ I've got black bars over the pages. In Tobit some content is Ok,

Hi List;

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

Hi, Looking to buy at least 20 license for g723. May i know where to purchase please? Regards, Nhadie

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel

Hi all, I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? Which licence could I use and how many ones are required (only one per phone or also for voicemail and MOH)? Regards Amaury -------------- next part -------------- An HTML

Hi everyone, first of all, I searched through the archives for any posts resembling this, I didn't read all posts about the patents so if this has been suggested before I apologise. I read in an article on C|Net (I think, it was linked from Slashdot anyway) that Thompson are threatening to sue you if Ogg Vorbis becomes a success. Which is evil, and I'm also mad at them because they never

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