similar to: Channel banks, voicemail, and immediate=no

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From:

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

Currently number of error reporting register banks is hardcoded to 6 on AMD processors. This may break in virtualized scenarios when a hypervisor prefers to report fewer banks that the physical HW provides. Since number of supported banks is reported in MSR_IA32_MCG_CAP[7:0] that''s what we should use. Signed-off-by: Boris Ostrovsky <boris.ostrovsky@oracle.com> ---

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Where can I find a good tutorial on how channel banks work? I need to get a 6 port (or so) channel bank for FXO. I need to find some information on which ones are supported well under Linux and with Asterisk, how to configure them, what specifically to look for in a channel bank, etc. I'm pretty new to all this, so I'm not familiar with a lot of the terms and such. Thanks! Steve

I'm looking to buy a channel bank. I'll be using this to basically dual a Toshiba PBX and Asterisk so that we can deploy Asterisk in small steps and have the Toshiba as a fallback in case something goes wrong. What I need is a channel bank that will allow me to take in 12 lines using the FXO ports in the CB and then output those lines to the Digium T100P. Has anyone had experience

Hi all, I'm currently evaluating ideas for a phone system for our new office and I have a copy of Asterisk along with a couple of H.323 phones setup in the lab here. I'm very impressed - thanks for the great software! (I was especially impressed that the IAX connection to Digium worked out of the box like that - someone at Digium no doubt has a voicemail from me this morning!) ;-) What

Hi guys, I'm new to the telco game and still pretty new to Asterisk, although I've been using it for a couple of months now and like most of what I see. At my office, we've got a small two extension setup w/ two Digium cards for a single FXO line and three FXS extensions, but I'm also working on designing a larger installation for a customer which will involve ~16 analog

hi, i have some newbie questions about channel banks. i have an adtran act-1241 sitting around. it accepts D4 modules, and it contains a number of e&m cards. first of all, how does this thing work? a t1 contains 24 channels, and i noticed that the channel bank has space for 24 cards. what do these cards do? what are their outputs? the ones that are in there have some outputs on the front

Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. Deepak

Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are

Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card.... We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Hi I am using DISA to get my Polycom SoundPoint400 with H323 firmware to connect to * I have it working, but when I dial SIP end points there is no ringing tone on the phone. DISA gives dial tone but does not give ringing (if I understand correctly it is because it expects to transmit sound created by terminating side of the call) Is there a way to make DISA application to generate ringing

U-boot for the arndale board splits the memory in 8 contiguous banks and rewrites the memory node. So most of the memory is lost. As the frametable is only able to handle contiguous memory, use the first contiguous banks and warn if some of the memory banks are not used. Signed-off-by: Julien Grall <julien.grall@linaro.org> --- xen/arch/arm/setup.c | 23 +++++++++++++++++------ 1 file