similar to: simultaneous ring - call groups or queues or something else?

Hi: Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server isn't seeing the mainboard's APIC. -Stephen-

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

On Sat, 2007-07-07 at 08:39 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Fri, 06 Jul 2007 12:02:53 -0600 > From: Stephen Bosch <posting at vodacomm.ca> > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID:

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

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

Hello, I'm following the thread "Asterisk@Now vs Trixbox", and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom "step by step" installation on a distribution of his/her choice? Thanks Stefano

For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general]

How do I manually set up a ring group? All the info I've Googled tells me how to do this using Trixbox or FreePBX. I am using standard Asterisk 1.4 configuring at the CLI. Michael

Hello I installed Trixbox 2.0rc1 at home and subsequently upgraded to the final release. The problem I'm having since upgrading it to 2.0 is that when the system receives a call on any of the trunks, the caller doesn't hear the ring back. This means: if I call my house from my cell, the phones ring at the house but I can't hear anything. Any ideas? Thanks Marcelo

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere

Hi, In my Office all our users have a Desk phone. Some of the users who are using laptops have a Soft phone too along with their Desk phone. Right now we are using two different extensions for their desk and soft phones. Is it possible to have simultaneous ring for both the extensions (ie. soft phone and desk phone). I tried using " Dial(SIP/desk & SIP/soft) " and it works fine

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings

I have a client who is looking at replacing their PBX, and I'm interested in putting together an Asterisk solution for them. One feature that would really, really get their attention is if I could do the "Vonage" thing, where if a PSTN caller dials a direct extension (coming in over PRI) both the user's deskset _and_ an external number (their cell phone) would ring, with

I have a need to issue a dial command to a number: same => n,Dial(${DIALGROUP1},${TIMER1},t) After a number of seconds, let's say 10 seconds. I want to dial another set of numbers while continuing to ring, or interrupting the first group of numbers. same => n,Dial(${DIALGROUP2},${TIMER1},t) Is there a way to do this without interrupting the first call? Thanks, Scott Haley If you

Hi Folks, Struggling with a new * installation with 2 Rhino R2T1 cards. For some reason, the system is locking up tight when you run ztcfg to configure the card(s). Configuration is asterisk 1.2.14, zaptel 1.2.12, and rhino's 1.05rxt1 drivers. The cards seem to load fine with a "modprobe rxt1", but once you run "ztcfg -vvv", the system will lock up within a few seconds, no

Hello Everybody I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

I am seeing the following in my log file (standard trixbox install). One seems to be complaining about an error in the dialplan but it won't tell me what file or what line. The other (maybe related) is complaining about a channel lock. How to do go about trying to figure out what the problem is and how to solve it? ---------------Logfile-------------------------------------------- Nov 14

Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some