similar to: Setup up Intellitouch ITC-3002 Sip phones with Asterisk

I'm thinking of purchasing the Intellitouch ITC-3002 2line phones for a small 7person office. The documentation looks suitable for my needs. Any "Gotcha's" anyone run into? Thanks! Terrelle

Hello all, I'm looking to troubleshoot some echo issues and possibly "studdering" while using the speakerphone with the Intellitouch ITC3002. On the hardware end, i have a 2621 setup as the router with some policy maps for qos, and a 2900XL with cos priority set to 5. I have setup created 2 vlans currently (really small office), 1 for corp/data, and 2 voice vlan. Asterisk is on

Hello folx! I just started to play with *. I first installed it this past weekend on my Solaris 9 ultra 5 test box. Now I'm attempting to put it on a freshly built Linux box a mere few hours old. I've installed asterisk 1.2.7.1, libpri-1.2.2, zaptel 1.2.5 and the latest asterisk-sounds 1.2.1. This is running (or installed) on my Slackware 10.2 box running kernel 2.4.31. I'm testing

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

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Hey guys, I'm actively trying to get the "big" picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and

I am trying to implement a Centralized Call Waiting System. I have red some document about asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource.

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent:

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

I'm trying to get an Elographics serial touchscreen working with CentOS 6.2 using the xorg-x11-drv-elographics driver. Touching the screen will wake it up if the screen has blanked, but it never moves the cursor on the screen. There is also a keyboard and mouse connected while I'm trying to get this going and they are loaded automatically and using the evdev driver. Here are the relevant

Hey guys! I'm the past week and a half, I have really learned a lot from the mailing list and the wiki's posted online. Now I have a question regarding different ways I can setup my asterisk server for a small business with 12 extensions in the office. Cost is a great concern, so I know cheap analog phones at the desks is what we are looking at. My question is, should I go do a fractional

Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a "production" machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip):

Hey Guys/Gals, I was thinking of getting a cisco 2621 router with 2 FastE modules and 1 T1-Wic to use for a small business setup along with asterisk. I want to provide redundancy between my VOIP provider and telco, or between two VOIP providers. Also my plans are to setup VLANS to separate internal voice traffic and regular data traffic using a cisco 2900 switch. Is the 2621 overkill for 5-7

Any one know of some "stores" that I can walk into to buy some VOIP equipment? Maybe some Phones? I know there are website I can hit, but I want to actually hold some equipment in my hand and ask some questions before purchasing. I'm in the Los Angeles area. Thanks! Terrelle Shaw

Hello guys, I'm at my wits end.. hope someone can help.. this is the deal. I have setup a linux box to serve as a "gateway" for our internal lan. Running 2.0.34 slakware. I have 2 nic cards.. One for our "internal" lan (192.168.2.0) and the "external" lan..I have gotten the OS to see both of them, but when trying to get DHCP to work with both, it kept barfing..

If you use a Sangoma T1 card, (A10x) card you can send both voice and data down the same T1 and have the Sangoma card split it for you. If you are talking about non-hobby usage, stay away from FXO adapters and go with a T1.. You'll be much happier in the long run. For a fractional T1, don't worry about hardware echo, asterisk will handle the echo fine. I would seriously consider shelling

On 11/25/05, Fernando Chmielewsky <ferchimi@gmail.com> wrote: > sorry I ask you this in this way... > I once tracked a bug in wine using CVS, but now I can't even get to > download the cvs tree, doing exactly the sama I did in the past: > creating ~/.cvspass with the single line > :pserver:cvs@cvs.winehq.com:/home/wine Ah<Z > and ~/cvsrc file containing: > cvs -z

Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use