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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.24s
  • Memory usage: 2.24MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 3.23

Digium

Based in high-tech Huntsville, Alabama, Digium is the creator and primary developer of Asterisk, the industry's first Open Source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures.

Digium solutions reduce the costs of traditional TDM and VoIP implementations through Open Source, standards-based software and next-generation gateways, media servers, and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, Loopstart, and GR-303. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAX (Inter-Asterisk eXchange), SIP, MGCP, Skinny (SCCP), and H.323 VoIP protocols.

Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily Open Source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services.

Website: http://www.digium.com
Sales: sales@digium.com
Support: support@digium.com
Telephone Local: (256) 428-6000
Toll Free: (877) 546-8963
IAXTel: (700) 428-6000

Digium sells PRI & FXO hardware for PSTN connectivity and FXS phone jacks for Linux servers.
They also make the IAXy - a low cost IAX protocol 1 port FXS gateway.

The Open Source communications revolution is here, and Digium is leading the way.
The TDM FXO module uses the 3019 and 3050 chipset, where the 3019 PSTN line interface chip has many different PSTN line impedance settings including 600, 900, 270, 220, 370, 320, 275, 120, 350, etc, ohms. They also support Zcomplex impedance, as used in some International locations.

wcfxs.c supports international impedance matching in CVS HEAD from 25/06/2004

Job Openings:

Work@Digium

New Products:

Asterisk Appliance Developer Kit
Asterisk Business Edition Rev B.1

See also


Where to buy:




Created by oej, Last modification by willihu on Sun 29 of Jul, 2007 [14:45 UTC]

Comments

Comments Filter

Re: I can't receive incoming calls from the PSTN

by minka on Wednesday 21 of March, 2007 [13:47:23 UTC]
I have a similar problem, did you solve yours

Re: I can't receive incoming calls from the PSTN

by minka on Wednesday 21 of March, 2007 [13:44:16 UTC]
I have a similar problem, did you solve yours

Re: I can't receive incoming calls from the PSTN

by scilliant on Thursday 14 of September, 2006 [20:35:41 UTC]
The Digium X100P is a single port FXO card. FXO cards interface with analog line from the phone company. I guess what are talking about is FXS, which is not part of the X100P PCI Card design.

Good Luck

by scilliant on Thursday 14 of September, 2006 [20:31:04 UTC]

Digium Technology !!!

by scilliant on Thursday 14 of September, 2006 [20:30:10 UTC]
I witnessed some problems with asterisk performance in test lab. I work with scilliant.com and as we test out the execution outgoing calls (.call files) on an analog based Digium PCI Card (2FXO 2FXS). The asterisk server invokes the extension for the outgoing. call file. In the Scilliant lab the problem occurred as the extension execution started with the message file is played before a connection is established with the calling party. Another problem occurs when the asterisk stop all outgoing calls when you place large amount of .call files in the outgoing directory. The system somehow gets stock and needs for someone to restart the demand. I am not sure if anyone have came across these problem and is able to make the program detect these error states and do something about them.

I can't receive incoming calls from the PSTN

by omar on Thursday 29 of December, 2005 [17:02:12 UTC]
I have a X100P card, with Asteriks 1.2.1.
Everything is going well.
I can make outbound calls through the X100P, but I CAN'T RECEIVE CALLS from the PSTN.
Some one have the same problem.

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