similar to: voicemail+g729

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0 sources that I downloaded from Digium, my Polycom 300 works just fine. If I build with either various CVS builds, or the 1.0.6 sources from Digium, I get "No compatible codecs!". WTF? I'm using -the exact same- config files for both; I've tried enabling/disabling ULAW, ALAW and G279 on the

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256

I have Now a G729 codec license and when i start it comes: [format_g729.so] => (Raw G729 data) == Registered file format g729, extension(s) g729 [app_datetime.so] => (Date and Time) == Registered application 'DateTime' [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) sh: line 1: tmp: Is a directory rm: cannot remove `tmp': Is a directory

I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question.

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

Hello, I wonder if anyone else has noticed this. I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have a leaked file descriptor that remains until asterisk dies. Now, maybe no-one sees this, mainly because I have no g729 licenses on the machines where this happens. And conversely, I haven't yet studied servers that do have licenses. Why have codec_g729a.so loaded if

hi, How can I get corresponding shadowed page number of a virtual address in vm? -- Best Regards, Mohamad Rezaei

I'm trying to use the Digium suplied G.729 Codec, I have ran the register utility, and got my licenses written to /var/lib/asterisk/licenses, but when a start Asterisk I got the following errors: [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: This module is supplied under a

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk

I want my helpers to generate paths using a superclass instead of the subclasses. Assuming I have Owner and Member that both inherit from User, rails will use the current objects class name when generating paths: Let''s say current_user is a mod: <%= link_to current_user.name, current_user %> will generate "/mod/:id". I want to force it to generate

Hi, I've been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. I've removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a

Hi, Curiously, it''s worth noting that Xeon Processor 5100 series are not in the list of HVM Compatible Processors : http://wiki.xensource.com/xenwiki/HVM_Compatible_Processors Patrick ----- Message d''origine ---- De : Mohamed Zohny <mohamad.zohny@gmail.com> À : xen-users <xen-users@lists.xensource.com> Envoyé le : Mercredi, 8 Novembre 2006, 9h17mn 08s Objet :

Hi, We have a few PC with CentOS 4 installed. By default, it runs to runlevel 5, which displays the GDM. I'm just wondering, how do I actually make the GDM to start GNOME automatically (after a successful login) for all normal user account without having to add 'exec gnome-session' in every user's ~/.xinitrc? The current scenario is, only successful 'root' logins starts

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Hi I know with javascript we can validate the input of a form on the client, is there an easy way to specify this in rails? I am thinking the javascript code getting generated using some sort of DSL? -- Kind Regards, Rajinder Yadav SafetyNet Test Driven Development http://safetynet.devmentor.org -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: