similar to: identifying what a user pressed to reach my phone

I'll start by saying I'm a trixbox user, and a new one at that, so hopefully you can respond to me on those terms. I have a user who works from home 1 day a week. On that day I'd like for him to be able to connect with a softphone and be reachable by just dialing his extension as we normally would. I could set him up a new extension, then he could forward his phone there on

Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the "Allow Anonymous Inbound SIP Calls" box unchecked in freePBX. Is there anything else

I've read through the Wiki, I know its hard to nail down hardware requirements because it really depends on what you are going to do. I'm very new to Asterisk, haven't even read my Asterisk for Dummies book yet (its in the mail). Can you tell me if this sounds sane? We are planning on using a Dell 933Mhz dual CPU server, with 1GB of ram for our Trixbox setup. We will have 7-10

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later

When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes

Hi, For a 20 users prospective customer, I'm wondering if any GUI would allow and end user to edit an Asterisk IVR tree ? For instance, I'm looking for something allowing to edit interactions like : "wait up to 20 seconds and say this "to reach sales department, type 1, to reach tech support type 2" message" Regards -------------- next part -------------- An HTML

Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas?

Hi all, been searching & not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and

I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default Voicemail behavior. Standard behavior No answer/Busy -> send to Voicemail Requested behavior No answer/Busy -> message that if you press 9 you will instead be cent to reception -> send to Voicemail or Reception if 9 pressed. I want this to always happen when Voicemail is invoked. How

Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

Hi all, I don't know if this is the right list to ask, since I'm using Trixbox version 1.0.0.28, that has asterisk 1.2.17. I'm trying to configure the ring timeout value for my local extensions (when dialing from one to another), and the dial command simply ignores my values... I have one extension 0017 in my box, so I used the command Dial(SIP/0017|100|rTtWw) to dial to it. The call

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please. I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Metaswitch it goes to VM after a couple rings and the call goes to my

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists & irc & Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody

Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be

Hello everybody. First of all thanks to all the people giving their opinion on the subject I proposed: "Trixbox vs. custom install". You've all been very helpful. I try to summarize what has emerged from the various messages. Forgive me if I miss or forget something or if I simplify too much some of your messages... - Elman Efendiyev says that you should install from sources if

*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the

So I'm sure many of you are using or have tried to use TrixBox. Thus far, I'm in love with it. I haven't had a single snag. Then again, I don't need to get into anything overly nitty gritty with my Asterisk box. What are your views? -- Want a free copy of "TrixBox Made Easy"? Read the contest rules