similar to: Toshiba DK - Asterisk Integration

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=>parkedcallsexten => 4000,1,Dial(SIP/4000,60,tT)exten => 4001,1,Dial(SIP/4001,60,tT)exten => 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext => 700 parkpos => 701-720 context => parkedcalls [featuremap]blindxfer => # disconnect => *0

What happened to "sip show peers" in 1.4.13? Jerry

Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than

We have an installation with around 50 sip phones but only 5 of those are hardware. There are three Grandstream phones, one Snom and one PAP2T. We are running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are having problems which are either echo or distortion. The softphones all work fine and no one is reporting any problems. They are using 3Com switches which are fairly new. I

Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any

Hi all, i never understood that why is there 2 branches of asterisk going on parallel. asterisk 1.2.* and asterisk 1.4.*, i also heard about beginning of another branch which will be 1.6.*. so whats the difference between these 2 or 3 versions, can anybody plz tel me? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi everyone, Been reading up on Asterisk, and very interested in learning more. I've googled and read the archives and haven't found anything definitive on support for this phone system. We have a fairly large investment in the system itself and the phones, but would love to get away from the voicemail system it forces on us. Can anyone provide any feedback on using this system with

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. ---

> -----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

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

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

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

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

Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt

I set up a TE405P to go T1---*---Toshiba. I have the channels configured, and can place calls from the Toshiba, through * to the t1. Incoming calls work great to *, but if they go to the Toshiba, I get a hangup. I think the * is sending the call to the wrong span. I have 2 spans, span 1 from the T1, span 2 to the Toshiba. The bchannels show as 0/1 through 0/23 on both spans in * when it starts.

Hi, I am currently integrating our company's Toshiba PBX with the Asterisk version 1.2.1. I bought Quad T1 card, and making the port 1 to connect to PSTN PRI (use "pri_cpe" in zaptel.conf") and making the port 3 to connect to Toshiba PBX (using "pri_net" in zaptel.conf). The first stage goal is to just adding the Asterisk relay between PSTN and Toshiba system. The

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

Hey Everyone, Has any successfully installed this on a Toshiba A200? Any feedback would be great. Kind Regards Matthew Ollerenshaw Network Engineer * +612 1300 887 959 | Ê +612 9890 8349 | * mollerenshaw@visinet.com.au<mailto:mollerenshaw@visinet.com.au> | www.visinet.com.au<http://www.visinet.com.au/> Diploma (Network Engineering) | Microsoft Certified Professional ( MS