search for: mp202

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29 07:18:06] DEBUG[3218] chan_dahdi.c: Sent deferred digit string: T16w [Sep 29 07:18:07] VERBOSE[3...

...the best solution amen that is the less expensive solution. Another solution that i consider is mount asterisk on central office and IP PBX DIGISTAR preconfigured on remote offices. On the Users Side I was considering the use of Ata's or FXS Gateways, with Ata's I get offers of Audiocodes MP202, GRANDSTREAM HT 386 or Linksys SPA-2002. And with FXS Gateways sellers offers me Quintum Tenor AXG2400, Quintum Tenor AFG800 Thanks for any word that can help me to get this VoIP deployment working and sorry for my english. Cheers G.

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

Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian