similar to: Persistennt Data of Queue with Dynamic Agents

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

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

>recieve fax successfully. Today I tried to change to SpanDSP 0.0.3 pre 6 >but I just couldn't complie the app_rxfax and txfax application. The >SpanDSP 0.0.3 was successfully complied though. .3 is for developers only it is not intended for enduser use.

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta

Hi all, I an using MeetMe and I would like to use the -i function to annouce the join/leave of the user. However, this require that users record their names. Is there anyway to remove this? I just want MeetMe to annouce somethig like "A new user has joined the conference" and that need not to record user's name. Is there a way to do this?? Pim

hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon

Hello List, First, sorry for my bad English skill, I'm French. We have an asterisk 1.8.3.2 built from sources with a simple Queue : [TestQueue] strategy=ringall timeout=15 retry=1 timeoutpriority=conf ringinuse=yes wrapuptime=2 member => SIP/002E31,0,Agent A member => SIP/1CA3F2,0,Agent B member => SIP/E08972,0,Agent C And this dialplan (extension.ael) : 3600 => {

We've had a regular Park function in the past but recently I found the ParkAndAnnounce() application and I love the idea behind it. Here's a snip from the wiki (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce) so that we're all talking the same language: || ParkAndAnnounce(announce:template|timeout|dial|return_context) || || Park a call into the

Type "swift" at the command line so you can see the -options. Then modify the line to use the correct switch and specify the name of the voice you want to use. Thanks, Steve -----Original Message----- From: Pimjai Wesnarat [mailto:pw@nummerndirekt.de] Sent: Fri 4/21/2006 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] How

Hi All, I'm trying to add this function to my find-me application: when all available numbers are dialed in parallel , if any number is busy, take it at busy and go to voice mail. I read the Dial() Application but there's nothing written about this. My question is, is it possible to do this with Asterisk? Thank you, Pim

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.

Hello All, I currently have an Asterisk Box, running a callcenter with 04 queues. I set queues.conf with "persistentmembers=yes" in the general section as follows: [general] monitor-type = MixMonitor persistentmembers = yes However when I perform any kind of restart in the Asterisk application, all agents are considered unavailable after that. Though when performing

Hi, I'm trying to use AddQueueMember() to add a member to a queue and trying to make this "logged" member in the queue between reloads and restarts of asterisk. I configure en queues.conf: [general] Persistentmembers=yes And Extensions.conf: exten=> *01,1,AddQueueMember(queue_name,Local/${CALLERID(num)}@default,penalty); When I log with

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

Hey, I'm testing out some call queues. I have 7940's and 7960's with the SIP 7.4image. I have a queue that looks something like: [testqueue] strategy = rrmemory timeout = 15 retry = 5 weight = 0 announce-frequency = 0 joinemtpy = yes reportholdtime = yes I dynamically add a phone or two to the queue (AddQueueMember, not agents). When a caller calls in, connections are made and

Hi, I have a question about the definition of agents. The agents.conf file looks like this: [general] persistentagents=yes [agents] maxlogintries=5 ackcall=no wrapuptime=500 musiconhold => default group = 1 agent => 1311,1311,Tom agent => 1531,1531,Tim and here is the queues.conf: [general] persistentmembers = yes [queue1] musiconhold = default strategy = rrmemory servicelevel = 60

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]