similar to: Future installation questions - what do I need

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

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

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

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

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

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

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 have * installed as a voicemail only solution attached to a Toshiba Strata phone system. I am testing * at the current time, so it is not attached to ports designated as VM on the Toshiba. I am not sure if the VM ports would handle the call any different than an analog port. Currently if I call one of the DIDs that I have connected to the *, my greeting plays and I am able to dial my

I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

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

> > > 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]

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

so, the rumor is that apple will announce an authoring-tool for e-books tomorrow, and most specifically for digital text-books. and, if you look closely, they _are_ rumors. nothing more... plus, they don't even seem to be coming from sources inside of apple. nonetheless, the rumor is "authoring tools". "garageband for e-books", as ars called it... if it's true,

Hello list. I've been running a Cisco VOIP WAN for the last two years using CallManager V3.2. The remote 1760 routers are set up as H323 gateways. I've got a VG200 that is my PSTN gateway with a full T1 voice trunk to the local CO. I'm also running Unity v3.x for voicemail. We are currently using two ATA186 devices that are using the G.711 codec to pass incoming calls on the T1

Hello and thanks for any help provided. I have a windows xp desktop and a linux server, both behind linksys router connected to dsl with a static ip. I would like to run 2 nics in my server and route all local traffic through 1 and web host and mail server on the other. My question is this: Is this really practical? I spent a week already trying to get it to work to no avail. What would your

Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or

So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? I built another route 01144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest

>Message: 9 >Date: Thu, 02 Dec 2004 11:33:16 -0700 >From: "Kevin P. Fleming" <kpfleming@starnetworks.us> >Subject: Re: [Asterisk-Users] Polycom 500, asterisk user opinions? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <41AF5FEC.7050406@starnetworks.us> >Content-Type: text/plain;