similar to: How do I pick up a specific call from a queue?

We have a QuadBRI ISDN card from Digium. We would like to make it work with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of bristuff from the Digium homepage fails, both the stable version with asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD. Has anybody here succeeded to make this work? Or could we even be so lucky that somebody made RPMs for this? Lars

I would like to know how to install asterisk 1.0.9 with zaphfc working on a SuSE 9.2. I tried this: - The rpms with SuSE 9.2 are asterisk 1.0.6 - bristuff works, except for zaphfc, which doesn't compile. - The official asterisk download file doesn't contain isdn bri support Any ideas? Lars Dybdahl.

Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

hello! i'm trying to implement a callback feature. to accomplish this, i've written a python script(callback.agi) that starts another script as a independent process(with spawnl), without asterisk waiting for the other script (callback_dead.sh) to finish before it goes to the next extension. running it on the commandline seems to work, the script starts the other script and

Hi, I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2). I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I

Hello, I have a question about the lars package. I am using this package to get the coefficients at a specific LASSO parameter s. data(diabetes) attach(diabetes) object <- lars(x,y,type="lasso") cvres<-cv.lars(x,y,K=10,fraction = seq(from = 0, to = 1, length = 100)) fits <- predict.lars(object, type="coefficients", s=0.1, mode="fraction") Can I assign

There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade. There is an issue while a call goes to any queue we create, the call is being disconnected after 20 seconds and it is hangup. The following is the configuration: - vi /etc/asterisk/queues_additional.conf [8] wrapuptime=0 timeout=30 strategy=ringall servicelevel=5 retry=4 reportholdtime=No queue-youarenext=

Hello, I've been looking for this on voip-info and this list threads, and I am guessing I am not looking right. What I need is the way to capture (and write to DB) the information on who 'picked' or 'received' the incoming call. Here is the example of .rb file that is called from extensions.conf: private def lokal

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

I have an app that generates callfiles in the outgoing queue, which connect a channel to an AGI (Perl script) at an extension. The AGI calls the Dial command over a SIP channel. Sometimes I need to stop the outgoing calls after the requests have been made. I delete the callfiles from the outgoing directory, but there are still some calls "in the pipeline". Especially if Dials failed at

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

I have been successful using ChanSpy on a standard Dial call but when attempting to ChanSpy on an incoming call that has been added to a MeetMe conference (attempting to coach an agent that is speaking to a conference of callers) it seems to fail to connect to the channel. Here's the console dump: -- Accepting call from '2154182700' to '3399' on channel 0/18, span 4

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

Hi folks, I've got trouble doing an blind transfer from an "EXEC Queue quename|t" in an AGI script. Attended is working fine, also when doing the same queue from the extension.conf file is fine. Here's my log; -- Executing AGI("IAX2/utv01-5", "agi://localhost/queuecall.agi?queue=vxl") in new stack -- AGI Script Executing Application: (QUEUE)

I have a queue I configured as "strict" and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy).

Hi All, I am trying to implement ACD using Asterisk 1.2.18 and I've chosen AgentCallbackLogin for login purpose. One AGI is written which will actually get executed when agent dials '1001' (say) from his SIP phone and enters into the queue. Second AGI gets executed when the Dial operation is performed. I see the agi_uniqueid obtained from both AGI instances are different and I

Dear, I'm currently using a Dial command with multiple destinations and channels eg: Dial(SIP/100&SIP/101) I simply would like to know, in real time during the call (from dial plan or AGI), who has picked up the call. Can I find this information in a variable somewhere ? Thank you for your help Patrick

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for a queue up to 15 agents through a PRI line, it was working fine for more than 1 year, suddenly, when there is a load on the queue, the asterisk service disconnects and the calls are dropped. And the service starts again after few seconds, and so on. I am not using fax. I checked PRI by zttool and there are no alarms. The cdr logs