similar to: Transfer call help needed

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 am still testing Asterisk, but I am running in to a lot of problems. I set up numerous extensions, but Asterisk is not performing to tasks correctly. Here is an example. exten => 231,1,Dial(Zap/g1/231|3) exten => 231,2,Voicemail(u231) exten => 231,3,Hangup When I call in and enter extension 231, my call is routed to the correct extension, but it just keeps ringing. If I change the

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

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

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. ---

I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension 1111 picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to

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

> -----Original Message----- > From: Brian Roy [mailto:mister.roy@gmail.com] > I do this with a Toshiba Strata DK280. The model before the CTX. You > shouldn't have any problem doing this with the CTX either. Assuming > you don't have SMDI integration with your voicemail (you didn't state > that you did), you just need the phones to send trailing digits, and > you

> > > 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]

Hello all, first post, need help. I'm running a complex asterisk 1.8 install with five machines. I inherited it and don't fully understand it, nor the deep mysteries of asterisk either. I would appreciate any insight you might have. I scoured the 'net and the Digium wiki and my Google-Fu has failed me. I've been asked to somehow prohibit transfers to extension 3232. It has to be

>All, >I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and >are answerd by the auto attendant. When an extensions is entered I see >the Dial(SIP/100) on the console but the phone never rings... >I can pick up any extension and call 100 and it rings just fin>e and they >answer and everthing is fine. All 8 extensions can dial 9 for an analog line >and call

I am changing the title because this is really about the history of anova, and about strata in analysis of variance. As this kind of question has been arising very frequently, an extended comment may be in order. The ideas, and the sums of squares breakdowns, go back to Fisher; see in particular his "Design of Experiments", first published in 1935. This book is still a good

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an *@Home box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something

I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc

For those that have experienced low VM recording volumes when using a Digium TDM04b (or similar analog pstn card), a work around has been committed to cvs-head. Need some folks to test it; it doesn't seem to work for me, but need some feedback from others to ensure the work around is actually functioning. (The work around relates back to bug #2023, and was committed around Thursday or Friday

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the