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Groups for implementing =91GSM Gateways=92" which leads me to believe (or h= ope at least) that more than one phone can be paired to a dongle. Dongles are so cheap I guess it doesn't really matter other than more complexity. Anyone know for sure? --=20 Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) --001636c92545598447046efba265 Content-Type:

Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different

Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk supports SMS over GSM modem. I know chan_mobile had SMS in the future at one point but have not revisited the project since. "America Movil's MVNO TracFone Wireless quietly unveiled a prepaid, nationwide unlimited offering for $45 per month that includes unlimited text messaging and 30 MB of data."

Hi, Quick question to the real world. Approx what specs would I need on server to handle 95 ZAP or Dahdi -> SIP gateway using G729 on the SIP to carrier side (nothing else, just media conversion)? Does the latest Asterisk/DAHDI significantly improve these numbers over say, Asterisk 1.2.X? Sure, there is plenty to read but nothing I could find quickly on my exact needs that was clear and I

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

"app_rpt is an application which comes bundled with Asterisk, however, a later version may be available on our source repository. All you need to do is go to asterisk.org, download asterisk, configure the asterisk to build app_rpt by modifying the Makefile in the asterisk/apps directory, and then compile and install it. You can get the latest version of app_rpt.c along with the sound files,

I have an issue with one of my installations running Asterisk 1.4.20 that I need some help with. Not sure if this is new or not but Zaptel or Libpri is not releasing channels properly. I have had issues with calls that stay up for eight hours, long distance on the telco side, so it is more than just a nuisance. Does anyone know if there was a related bug in the 1.4 branch of Asterisk, Zaptel,

Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have

Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price

Hi, I guess I'd need to run Beronet quad and octo bri cards under bristuff to get zaptel features (echo canceling, timing source).... Am I right or could I achieve this also with chan_misdn - their native driver ? Running bristuff on Beronet cards is unsupported. Has anyone succesfully run Beronet quad BRI cards under bristuff recently ? Do they work ? Regards, Rob.

I'm not aware of any zaptel driver for such HFC USB modem (some Xorcom's products use USB, so ...) so I'm inclined to think it's not possible but it's better to ask ... Cheers ------=_Part_13548_28463665.1207585749504 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable Content-Disposition: inline Hi

Hi Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns). SIP ports mapped through firewall as we often connect from outside, but all SIP accounts have good passwords. However our telecoms provider picked up a few suspicious calls to places we do not normally call at times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected

Hi, Bristuff works great with HFC card... your compilation problem may come from your kernel configuration... You should check this doc, at least, for the redhat config : http://www.automated.it/guidetoasterisk.htm Then, installation of Bristuff works like as charm ! Bye David Masure -----Message d'origine----- De : vdasilva [mailto:aquamoon@wol.co.za] Envoy? : mardi 28 juin 2005 09:01

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Bourvine, > > So, why won't we save the big bucks we pay them, hire two professionals > (who cost less) and support an open source code by ourselves? This way > we depend on ourselves only. > > > > Thanks, __Yehavi: I remember hearing University of Pennsylvania have been using Asterisk for sometime. I am not certain where I came across that

Are there any step by step instrunctions on how to install drivers and I guess bristuff for this card? Just need to use it to handle voice on 2 BRI circuits (UK) then utilise with Asterisk and some Digium cards handling POTS phones (and some VoIP out the back). It's the EICON card stuff and how to make it all work I'm finding confusing? Steve -- NetTek Ltd UK mob +44-(0)7775 755503

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

I'm still trying to install a HFC-s BRI card onto *@home .6 I'm new to this so I probably am overlooking the obvious. Can I just install BRIstuff onto a fresh *@home install? The BRIstuff installer downloads another * from Digium. Will this interfere with the @home install and must I comment out the Asterisk install in the BRIstuff install.sh file? Any pointers are much appreciated.

On Saturday, April 15, 2006 3:17 PM Remco Barende wrote: > I heard that Junghanns is working on such an interconnection. It is > already possible to connect their PRI cards, and they are working on > BRI<->PRI. Correct. The next driver generation is supposed to support this fully. > I ise their bristuff for an HFC-S BRI card and am not happy at all > with the way they