similar to: Re: Asterisk-Users Digest, Vol 6, Issue 73

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

Adam, Tor sent this one a little while ago that looks really promising for solving the problem. Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tor Setane Sent: Thursday, January 06, 2005 2:09 AM To: Noah Miller Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re:

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

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

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, 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 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 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

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 don't know if Polycom decided to change their policy or not, but I just went to their website and when you click on the downlooad link, it will now present you with an option to join their extranet. I did that and I got the following message immediately: <snip> Access to certain Resource Center sections and content are subject to approval. Eligible members, will receive further

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

>Message: 10 >Date: Wed, 26 Jan 2005 17:53:39 -0500 (EST) >From: "Sean A. Newton" <nester-asterisk@wewt.net> >Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again.. >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: >

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

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

I have the Polycom 500 and 600 phones. Rather than put an entry in for each line appearance, I would like to use the feature that shares one extension for the lines, so that I will get the call on the enxt available button. How do I configure that? Chris Mason

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 - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use