similar to: [Asterisk-Dev] Digium's G.729A codec problem

Hello, I would like to know if FreeBSD has drivers for TE405P cards submitted to the ports. I have acquired a card and would really like to use it with FreeBSD. The wiki said the drivers havent been tested yet. Have there been any updates since then? regards Kavit _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com

Hey everyone, I posted bug #3746 for people to report on the latest code in the stable branch. Once I get enough reports that it is working fine, we will release 1.0.7. Thanks, Russell Bryant _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit:

Thanks Mark will try that out! -----Original Message----- From: MF Hulber [mailto:asterisk-admin@hulber.com] Sent: 22 March 2005 05:25 To: Reuben Grech Subject: [info] [Asterisk-Users] :: BIOS Motherboard Settings :: I have the same motherboard. I put the card in the 2nd slot from the bottom. In this slot, if you look at the manual, it will possibly be in conflict with some USB channels. I

For anyone using CVS HEAD, if you are using queue member persistence or agent persistence, your next update will cause the persistence to break. The storage format for these elements has been changed so that it can be more easily extended in the future, but this required breaking compatibility. This should be the last time these features will be broken by an upgrade :-)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the

Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in

After upgrading my small test system from Debian Etch->Lenny via a complete reinstall, I find my g729 hostid has changed. Same machine, same CPU, same NIC! It doesn't seem reasonable that I have to burn my one "no-hassle" re-registration for a simple OS upgrade. The README only says that hostid is based on MAC addresses of all NICs, but that doesn't seem to be true. Does

[ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source

Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

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

Hello! It seems that the new codec is not backward-compatible to glibc 2.1/2.2 so I receive the following error: [codec_g729a.so]Oct 22 02:37:18 WARNING[3433]: loader.c:325 __load_resource: /u sr/local/glibc/libc.so.6: version `GLIBC_2.3' not found (required by /opt/files/ usr/lib/asterisk/modules/codec_g729a.so) Oct 22 02:37:18 WARNING[3433]: loader.c:554 load_modules: Loading module

Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a soporte@avanzada7.com o a marketing@avanzada7.com

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

I tried to download Digium’s Opus Codec via the following link, but the server is unavailable: http://downloads.digium.com/pub/telephony/codec_opus/ It took me a while to figure this out, because initially I tried downloading via selecting the Opus codec in make menuselect and realizing that it isn’t there after make install step. Can someone from Digium/Sangoma please confirm? FLORIAN

I was told that my writeup lacked an example and details so I reproduced the code that X uses and I was able to boil down the issue to a couple of lines of code. Sorry again for the length of this email. Code was compiled on OpenBSD with clang 3.0-release. ======================================================================== With -O0 which works as X expects:

I'm getting an INVITE from the Cisco softswitch looking like this (from sip debug trace): Sip read: INVITE sip:9043940358@206.165.120.52;user=phone SIP/2.0 Via: SIP/2.0/UDP sia-ATLCA146.telefyne.com:5060;branch=z9hG4bK_1146_38l5 From: <sip:7278673170@sia-ATLCA146.telefyne.com;user=phone>;tag=1_1146_f153592_732 j To: <sip:9043940358@206.165.120.52;user=phone> Call-ID:

Hi All, I want to use Asterisk for VoiceMail for a softswitch. I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0 Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal.

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to