similar to: Softphones and other VOIP PBX's

I've been trying all day to reach their techie folks to ask a couple questions about something I worked all weekend on. Keeps rolling to VM and the receptionist does the same thing. Just was wondering if anyone else was able to reach them today. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax)

I suspect if someone recompiled the module that handles the transcoding it could take advantage of say a AMD64 bit cpu. Although I bet cache has a big impact on performance with this. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Monday, January 10, 2005 4:20 PM To: Asterisk

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I

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I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

Hi, Does anyone know what is going on with voipsupply? My sales guy hasn't been online in several days, their 800 number is fasy busy, as are their direct lines. And the canadian store website is down. What the heck is going on?

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to

Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone.... Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com

I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much

Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

(Resending since the archive seems to have eaten the text.) I'm working on getting Speex running on a TI Davinci (6441), on both the ARM and the DSP. My immediate goal is to understand the processing requirements on different architectures so that I can properly specific an engineering project. I have it up and running on the ARM, and I just wanted to see if anyone could sanity-check

System is built on a SuperMicro motherboard with Serverworks chipset, IRQ is not shared. Have a dialplan that worked for 8 months without errors, tried reverting to older release then upgraded to 1.0.3 stable release, currently running on fedora core 1 kernel 2.4.22-nptl.2199 (have tried plain jayne), telco says "it's not us", HDLC abort seems to occur when when a Zap channel hangs

Dear all, I wanna know what can I do to get the PBX's clock from -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090428/3218b8b0/attachment.htm

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

We are looking to find someone that is familiar with Fujitsu and Mitel PBX's. Email russc at inline.net off list. Thanks

I have a client which is willing to connect * with a siemens hicon PBX. Any experiences that you wish to share, including FX, ISDN or E1 connectivity issues? PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030707/dfb33d1f/attachment.htm

Hi, We're using our legacy PBX as a channel bank with asterisk sitting between the pbx and our telco provider spliced by a TE410P. If it were a straight analog FXS card then we'd use a hook flash to break into asterisk for transfers etc, does anybody know what the equivalent is for the PRI zaptel support? Regards Steve Steve Hanselman Brendata (UK) Ltd Tel:

A friend of mine runs a small office building, 10-15 tenants. Each have their own company, their own thing, renting space from him. His main PBX is getting dated and his tenants are complaining. I was telling him about Asterisk but his main concern is he doesn't want to have to always be the one to add/remove extensions, or change the IVR hours or whatever. Does anybody know of a free or

Yes, try http://1ezphone.com its a browser softphone. ----- Original Message ----- From: Zoa To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] SIP Softphones and Citrix ? Date: Fri, 01 Feb 2008 23:09:56 +0200 I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: >