similar to: sip, call ransfer and call waiting

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

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'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

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

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 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

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

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

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

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. > > >

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

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

Has anyone been able to find a way to disable call-waiting on Polycom phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/b7ea7b5d/attachment.htm

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,

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

Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE

Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com