similar to: Asterisk, Zaptel and Legacy Phones?

Hi All, Has anyone had experiences with Asterisk and ISDN Gateways or cards? I am looking to setup a home phone system using asterisk and am looking to convert my two existing no's to a single isdn2e line. I was wandering if anyone could recommend which is better to use with asterisk - ISDN Card or gateway? also which brands are best with linux/asterisk? I have used eicon BRI cards for

Hi, I think this is one area that needs to be developed. I am curently implementing a system for my home so cannot really justify the cost of financially supporting the development of this when all I really need to do is buy a telephone extension lead for my existing fax modem!!! I am more than willing to devote some testing/documentation time (I am not really a programmer) if that helps.

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

Hi All, I know that this may not strictly be an asterisk issue, but if anyone has any ideas on this prob it would be appreciated. I have a working asterisk setup using Suse 9.1, a BT ISDN BRI card (Fritz?), sipura 3000 and a few cisco 7940's. I wanted to do add faxing into the equation, so I thought I would try and setup the following: PSTN->asterisk->hylafax->pc At the moment

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

I have tried sjphone - worked well, although I think my 3 year old IPAQ had a bit of a hard time keeping up with the pace as there was quite a delay in the speech. Probably says more about my ancient IPAQ than SJPhone. Sam Lex Lethol <lethol@gmail.com> wrote on 23/09/2004 15:31:39: > I tried the xten one and didn;t like at all.. > > Havent tried to SJPhone, but my guess is

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Does anyone know if this phone can or does work with Asterisk ? I have a potential client who wants to throw away their current mitel ip system (keeps crashing on a regular basis - not good for a phone-based company), but are reluctant to purchase new handsets. Thanks. Julian.

hi all, i have problems running this code (taken from: http://blogs.msdn.com/michkap/archive/2006/12/20/1332470.aspx) in wine (1.1.26). the GetKeyState trick is not working, so GetKeyboardState never fills in any information in windowless apps. > /* > * If this thread needs a key state event, give one to it. There are > * cases where

Hello everyone, I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line="2 WCTDM/0/1 FXSKS" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by

The Wine development release 1.7.41 is now available. What's new in this release (see below for details): - More Known Folders supported in the shell. - Some more support for kernel job objects. - More MSI patches improvements. - Some theming fixes. - Various bug fixes. The source is available from the following locations:

Hi, I've been looking into the one bad thing about * which is there's no practical solution to running a console. You know the kind where you have rows of buttons each representing an extension. You press the button of the extension you want to transfer the call to, and it's done. There's the beginnig of GUI version but it's going to eat resources for running X which

I've been trying to cut down the size of openssh so I can run it on my Nokia 770. One thing which helps a fair amount (and will help even more when I get '-ffunction-sections -fdata-sections --gc-sections' working) is to have the option of compiling out privilege separation... Is it worth me tidying this up and trying to make it apply properly to the OpenBSD version? Does the openbsd

gReetings: Is there any way to download a (or a sample of a) crossection of stock market prices? Or is it possible to use get.hist.quote with a *wild card*? Thanks, mihai Mihai Nica 170 East Griffith St. G5 Jackson, MS 39201 601-914-0361 ____________________________________________________________________________________ 8:00? 8:25? 8:40? Find a flick in no time [[alternative HTML

Hello! Please consider including the attached patch in the next release. It allows one to drop privilege separation code while building openssh by using '--disable-privsep' switch of configure script. If one doesn't use privilege separation at all, why don't simply allow him to drop privilege separation support completely? -- Sincerely Your, Dan. -------------- next part

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

Hi All, Here at my office we have two basic ISDN Access with two phone lines each. We would like to install an Asterisk PBX. The question is: How can I connect the ISDN lines to Asterisk? Is it possible to connect to a DIGIUM Wildcard TE410P card? Or this E1 card is only for PRI access? Any other suggested and tested hardware for a production box? Let me know, -Z --

When I installed my first home-PBX three years ago, I was looking at "cellsockets" -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a

Has anyone else had issues with upgrading from SIP 2.2 to SIP 4.4 on the Cisco 7940? I'm following the directions outlined by Cisco. Is there a trick that I'm missing? Clues welcome.

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris