similar to: G723 pass-through and codec negotiation

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Thanks...which version of IPP did u use ? I do not have Makefile file....there is only a .sh script Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Juan Salas Inviato: luned? 3 ottobre 2005 15.41 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE:

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 sequence I have two

Can anyone recommend a codec pack with G723 for use under Vista? I have G723 prompts (about 70 prompts totaling 1MB) needing to be converted to G711 uLaw. I tried Audacity but it doesn't have G723 codecs. I tired some google found adware free tools and websites with no success in converting G723. It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD) can do it -jason

Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've a Cisco here and it works fine with G723, but not with my asterisk. The bandwitdh is very important, since we will have our extensions at home. I know that I have what I pay, but the phone works with cisco. Trying to use G723 or G729 Asterisk says no codec available. Does anybody have it working with any compression

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan

Hello I had this codec negotiation with oh323 call. i used G723 codec and the provider had G729 as first priority. In this situation what ever number i dial i used get "No one there to answer the call". As soon as i changed my codec to G729 the call went through but had other problems, which i got away by dowloading the latest code for oh323. Has anyone seen this problem? or it is

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080424/b442d5af/attachment.htm

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Hi, just a question: anyone has never installed g729 codec on VIA motherboard with C3 processor ? I'm having problem with IPP libraries, and Intel said that it works only on Inter processor. Any suggestion? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

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

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

El Tue, 9 Jun 2020 09:46:32 -0600 George Joseph <gjoseph at digium.com> escribió: Hi George > > > > > > If transcoding is enabled Would it be possible to do the same but handle a > > 488 > > back from Bob and failover to another INVITE with Bob's allow list to > > handle > > transcoding? That way we would always try no-transcoding before

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

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

hi, hi we have G729 codec from Digium, without the G729 codec, we can do the hash transfers to other sip phones fine. but once we are using the G729 codec, the asterisk is not responding to hash transfer, ie, when we press "#" it does not detect it and says "transfer..", is this a problem with G729 codec? (for testing purposes we have bought licenses for 2 chs) this also

Ok, My provider is sending a call to me via ULaw but Asterisk isn't picking up on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my sip.conf and that's the only thing I allow, but when my provider sends me the requests, I get an error about No Compatible Codecs: 17 headers, 8 lines Using latest request as basis request Sending to 67.19.245.213 : 5060