similar to: Inserting white noise / music / sound file into mixmonitor

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Is there an equivalent stopmonitor command if you are using MixMonitor ? StopMonitor does not seem to have an effect on MixMonitor Julian.

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part.... exten => *0,1,NoOp(${PQMSTATUS}) exten => *0,n,Macro(user-callerid,SKIPTTL,) exten =>

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.

Hi All, I have just started looking at the code for Xen so please bear with me. A domU Linux kernel running with independent_wallclock=1 seems to sync its time with dom0 after every "xm unpause" (obviously preceded by an "xm pause"). I don''t see where the xtime variable is being updated after an "xm unpause", i.e., domain_unpause_by_systemcontroller().

I set up Alpine Linux as a libvirt-based KVM hypervisor last week. Everything was working beautifully on Friday. I left two Windows VMs up and running and two 'Saved' on Friday afternoon. When I go in this morning the two running ones were 'Paused' and I can't unpause them with virt-manager or with virsh. The other two saved ones I can start, but they are immediately paused

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

I run a centos6.2 guest VM in xen4.1.2, the id of centos-domU is 22. The info of dom0 is: 2.6.32-131.21.1.el6.xendom0.x86_64 #1 SMP Tue Dec 13 18:09:29 EST 2011 x86_64 x86_64 x86_64 GNU/Linux Centos-domU: 2.6.32-220.7.1.el6.x86_64 #1 SMP Wed Mar 7 00:52:02 GMT 2012 x86_64 x86_64 x86_64 GNU/Linux At dom0, I went like this: [malei@xentest-4-1 Tue Apr 30 ~]$ sudo gdbsx -a 22 64 9999 Listening on

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back.

I'm trying to implement a virtualization API. I was testing migration with libvirt I got some problems. When I use the following command : *virsh migrate --live --persistent --copy-storage-all vm-clone1 qemu+ssh://server_ip/system* the migration works fine but in the destination host the migrated vm is paused and I can't unpause it and I need to reboot the vm to be able use it in the

Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a