similar to: Has anybody experience with SetGroup / CheckGroup commands?

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 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

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

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

>From this list I found that I could use SetGroup and CheckGroup to do what I wanted. But I'm not quite sure how I do it. The case is that I have 3 user groups, and one main group. The main group is for all the incoming calls from external phones. The main group should be allowed to have 3 calls at the time. The 3 user groups are internal groups that I want to limit by ony having one

Hi all, I am just scratching my head trying to work out a way to use SetGroup to check busy status on a sip to sip call. The complication is that one call can't be in two groups so I have got no way of setting busy status on both the calling and called party. Has anyone got a way around this. Thanks Daniel -------------- next part -------------- An HTML attachment was

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. > > >

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

How do I limit the length of an extension? In my test IVR/Automated Attendant (whatever it's called), at the beginning it plays "if you know your parties 3 digit extension, you may enter it now) and then it gives a list of options. If the caller puts the 3 digit extension, it goes through fine, if they press 1, or 2 it goes to the selected menu option, but if they dial 91235551212 it

Hi John, Kevin, Tor and Wiley (and everyone else) - >> I guess the phone just doesn't register as busy when there is only one >> call on a line. It has to have two calls on a line appearance to >> register as busy. Has anyone figured out how to disable this hold >> feature and just have the second call go to the second line, the third >> call to the third line,

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

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

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

Hi all, We've got a department that has 5 phones using a * 1.0.9 box. They need to have an extension that rings all 5 phones at the same time. Getting all of the phones to ring isn't a problem, but they are running into a problem with the phones ringing in their ears when they are already on a call. Example: Caller one calls the queue, all of the phones rings, and employee one

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

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

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

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

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