similar to: openSUSE Asterisk Packages

Hello everybody, i am searching a solution for a videoconferencing, Any solution (Free/commercial). Asterisk is a great software, but recently we have more and more demands about videoconferencing of 3 or more peoples, Existing solutions are heavy and costly, around 2500? for 1 client. This is insane. Is there any solutions out there for non millionaires ? Or even Free ? I remember a company who

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:

My equipement : 2x tdm 400p /4FXO/4FXS 16 PORT ASTRIBANK My first TDM400p is the sync master, so i set astribank to sync to it, but the quality is bad, like it is going fine and second after >>ROBOVOICE ;) any other devices (isdn/sip/tdm cards) are working fine, looks like astribank dont like to be sync 'slave'. Is there any way to change sync master manually, so every another

Stop advertising. Le 26 sept. 2010 09:46, "Faisal Hanif" <faisal at vopium.com> a ?crit : Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AMI. I have tested it in JAVA and it

Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen -------------- next part -------------- An HTML attachment was scrubbed...

Hello guys, Nouveau's OpenSUSE Build packages have broken dependences. It seems that now it demands xorg>=7.5 but the avaiable in opensuse 11.2 (current version) is 7.4. As there is no xorg-7.5 compiled for 11.2, I'm running vesa driver by now. I guess that a hint on download page could be useful. Cheers, --- Luiz Angelo Daros de Luca, Me. luizluca at gmail.com

I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download

On Mon, Oct 08, Richard W.M. Jones wrote: > On Mon, Oct 08, 2012 at 08:50:47PM +0200, Olaf Hering wrote: > > On Mon, Oct 08, Richard W.M. Jones wrote: > > > > > But out of interest, how are you creating this appliance? I thought > > > we'd ripped out support for old-style appliances from the rest of the > > > toolchain ... > > > > I use

Does anyone have a copy of spc.exe they could send me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

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

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

Instead of using a PCI card is it possible to use an outside SIP service for "CO" lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/33b131e3/attachment.htm