similar to: Dial command problems

I currently have a Toshiba Strata DK424 with a Stratagy voicemail system (4 ports). I am looking to go from having a receptionist answering the phone to an automated attendant. It appears that Asterisk can be the solution, but I have some questions. Do I just replace the Stratagy with the Asterisk or do I need to reconfigure the routing of the phone lines? Currently, the POTS lines come in to

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

I'm posting here in case anyone else runs into this and needs some help. I'll probably update the voip-info Wiki pages on Toshiba integration in a bit. Asterisk 1.6 makes things a bit easier than what is on that page. I'm integrating an Asterisk server with a Toshiba Strata system at my office. Right now, it is driving some VoIP phones (Cisco ATAs with analog phones plugged into

We've done a direct swap of an old Amanda voicemail system with a shiney new Asterisk system (Asterisk 1.0.9). The system consists of 4 FXO ports on the * box (TDM400P), and three old Wildcards we aren't using (too buggy we found). CO lines-> Toshiba -> FXO ports on * We want to branch out a little more and use it as an auto-attendant. The first problem seems to be an asterisk

I found some postings from Google (notably from Mark Spencer) about successful integration of a legacy Toshiba Strata system and Asterisk. I am also facing that current dilemma. The general legacy solutions that I can come up with is very easy -- either making Asterisk a "proxy" (or frontdoor) to the Toshiba system, or have it operate behind the Toshiba via regular extensions. I'm

I'm interested in setting up Asterisk as a voice mail system for our Toshiba Strata CTX100 PBX. Our current voice mail system connects to the PBX with four analog extensions. Does anyone have information on how to integrate this kind of system with Asterisk? I've looked at the legacy integration section of the Wiki, but it doesn't seem to have anything about Toshiba systems. ---

Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the

With some carriers the initial Audio (2-4 secs) seems to get cut off when using a Auto Attendant or Conf Meetme. Is there any known remedies for that. Just want to know if others have seen that esp. with Level 3. If Auto Attendant says - "Welcome to ABC bank" Caller only hears "Bank" -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have googled, and wiki'ed until blue. Is it possible to put T1---*----Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf [channels] switchtype=national context=from-pstn signalling=pri_cpe usecallerid=asreceived echocancel=yes echocancelwhenbridged=no echotraining=400

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity

Hi All, I''ve got a DomU that sometimes goes mad. I can''t ssh or usually even console to it. The time I did manage to console I got a load of dumps about being out of memory and swap, but couldn''t run any commands to find out which process had gone mad :( From Dom0 I can see the DomU at 100% CPU and can only stop it with a destroy. What can I do/check to find

> > > Hi, > > > I checked by changing to from-zaptel, but no luck yet. Pls guide me on > this. > > Regards, > vudura senadeera > > > ------------------------------ > > > > Message: 9 > > Date: Fri, 19 Jan 2007 16:47:18 -0000 > > From: "Robert Jenkins" < raj@jrw.co.uk> > > Subject: RE: [asterisk-users]

Hi all ! If someone could give me a hint on how to proceed with the following i'd be very happy: I have a test setup on an nrf52832 (Cortex M4) in which I receive audio from a PDM microphone (64 sample frame) and pass it directly to an I2S device i.e. from ISR to ISR. With uncompressed audio this works just fine. Now I try to insert OPUS1.3 in the path but cannot make it work. The

Hi all, I have a potential client interested in an Asterisk PBX, which will allow them to improve their virtual office services, among other things (it's a Business Centre). They currently have a Toshiba DK-40 and 19 DKT 2010-SD phones. The owner has told me that the phones are digital (not in the VoIP sense, obviously). Is anyone here familiar with this particular PBX? Any chance that it

Does anyone have any feelings one way or the other about running Asterisk on AMD vs running Asterisk on Intel? Thanks, Lyle J. McKarns ------------------------------------------- Networking/Linux Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax : 1 207 725 8552 Nexus Management,

I have recently changed jobs and I now have to administer a Toshiba DK424. I would like to integrate Asterisk with this Toshiba PBX. I am quite familiar with Asterisk but I have absolutely no idea about this Toshiba. To make matters worse the manuals for this system are missing. What I would like to do is place Asterisk in such a way that we can extend the Voicemail boxes and possibly add a few

With ESX and the vSphere Client, I can enable or disable a VM's NIC to "connect at power-on". This is very useful when cloning, since you can leave the NIC not connected, power-up the VM, connect to it's console, and modify ifcfg-eth0 etc, before bringing the VM online. Is there a virsh equivalent way of doing this. Thanks....Lyle

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hello, community. Several months ago Toshiba Inc. released a new communicator - "Toshiba Portege G900". All device's features are greate, but in practice things are going another way: communicator is really SLOW. Toshiba users have found that G900 is using nVidia GoForce 5500 Chip in it and it seems like it's working partially, because of an incomplete driver. We have written a

Hello all, Run groupbox.rb example in any of the two alpha versions and click the radiobutton in the upper right group box (those named Hilvesum 1, Hilversum 2, etc). You''ll notice that once a radiobutton is checked, it remains checked when some other radiobutton is checked. What''s the reason? Thank you. -- Best regards, Yuri Leikind