similar to: Limit incoming calls

Hi all, In our asterisk box we have a Fritz card and an HFC card. So far I've had the Fritz card working well, but its time to connect the HFC card up to the ISDN line and get it working. I've followed the instructions on voip-info.org here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install and here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install But when it comes to

I'm using a Grandstream IP phone to call someone through our asterisk pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses 2x ZAP-HFC cards. When I call someone, if the call isn't answered and then I hang up, I get "487" coming up on the grandstream phone. If I pick up the receiver again and then hang up, the PBX starts calling me back and when I

I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody. I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are pretty much the default, except in order to get the fcpci module to compile, I had to follow the instructions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install but the drivers would only compile if I left

I have a "Cologne Chip Designs GmbH ISDN network controller" in my asterisk box, and an ISDN line connected up to it. I downloaded the bri-stuff 0.1.0-RC4a package from junghanns.net and used that to compile asterisk using these instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install When I try to call my asterisk box, my sip phone rings twice and then the

We have an office full of budgetones and find the ringing volume to be too loud especially when all the phones are ringing at once on an incoming call. I've tried setting the volume on each phone but the volume setting doesn't survive through reboots or power offs. Has anyone found a way to make the volume setting stick? Thanks -- James Doherty - Systems Engineer Zeald.com - Websites

Hi All, Can someone please tell me how to limit incoming calls to SIP channels using the SetGroup & Checkgroup command. I don't want any call waiting on SIP channels and you are somehow meant to be able to do it with these commands. Many Thanks Daniel Niasoff

Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. Kind regards Cf --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

I am checking on the SetGroup / CheckGroup commands, but I have some troubles to undestand the examples. SetGroup(moh) can be moh anything as I like? Usually moh stands for "music on hold" CheckGroup(1) checks if somebody in in group "moh". Does it mean I can only have one SetGroup(xxx) ?? When I look at example 2 than I see two SetGroup commands and one CheckGroup

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the "on-hold" feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my

I have a rather long dial plan, but it includes support for call waiting. However, the setgroup checkgroup commands don't seem to be working. Can anyone help on this one? Excerpts are below. First exten-vm is dialed and then dial-new. As I understand, priority 1 increments the active channels for the caller and then in "dial-new" priority 8 increments for Arg3, or the Callee

Hi, I have been trying to get my * box to limit an extension to one line for either an inbound or outbound call anyone got a quick example I can look at or a good howto? Cheers, Dee

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

Does anyone know if SetGroup and CheckGroup apply to only current context or is it per server based? Ta SJ

In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? "Sip show peers" shows me just if it is

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

If i get a 8XX number, my provider told me that they will send all the calls he gets. But due to bandwidth and asterisk capacitiy in this particular installation, the system is only able to handle 27 incoming calls. How in my dialplan do I regulate, sending a busy signal, when my system hits 27 incoming calls? thanks, -- ------------------------------------------- Erick Perez

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are