similar to: Detecting hook flash in asterisk

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

Problem resolved. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back. extensions.conf [context] exten =>

Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the

Hi folks, After a fairly extensive Google trawl, I don't think the following is possible but would appreciate confirmation from anyone else who has tried something similar. I have an AGI (not particularly relevant) which is executed when someone calls into a specific extension. This AGI finds a suitable 'agent' (not actually a queuing system in the Asterisk Queue sense) and Dial()s

Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register extension 100 but with wrong password, so registration failed; 3. A then showed it was unregistered! Failed registration of device B shouldn't kick A off, I expect A stay

Hi, I have a windows executable which I am trying to run in Linux using wine. When I execute the command : wine {ABSOLUTE_PATH}/file1.exe, file1.exe runs many other executables internally, like file2.exe, file3.exe and file4.exe. Now when file1.exe is trying to run the other executables, it is not able to get the path of the executables. I have the "PATH" enironment variable set

Hi there, I need a Python interface to asterisk manager for my own project. The voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+manager+Examples) lists 4 python projects for this purpose: Fats, py-Asterisk, pyst and StarPy. Because my project is rather small and I don't want to involve twisted in, the options left for me are py-Asterisk and pyst. So I want to ask your opinion:

Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from

Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

Hello all. I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO interfaces. When I call (or receive a call) from the pstn, I ear echo. This happens if I use a softphone or IP phone, and does not happens if the call is internal. Can you help me with this issue? Best regards, Vitor Flausino

I'd like to transfer all my pesky telemarketing calls to Jolly Roger . http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html In the middle of a call I'd hit some DTMF sequence, which would dial Jolly Roger and transfer the call after Jolly Roger answers. But blindtransfer requires an extension after you hear "transfer". And I don't

I am trying to use call parking. I have the following in features.conf [general] parkext => 700 parkpos => 701-720 context => parkedcalls When I try #700 from my softphone asterisk just passes it and doesn't interpret it. Can someone tell me what I am missing? I am using asterisk-1.2.17 Thanks, Alberto

Hi All, My client makes manual sales calls to prospects. He is often sent to voicemail on the prospect's side. If he finds himself having to leave a message, he would like to be able to press a key and let a pre-recorded message play into the prospect's vmail box. This is so he can maintain consistency in his message. Can anyone offer suggestions of how I could accomplish this

I want to be able to send a prerecorded message to the person I am calling. I know that you can use the application map to do this. Just to test I enabled the testfeature example that is in the features.conf file. When I hit #9 during a call the other user does not hear the monkeys, they only hear a series of beeps. I have tried with different soundfiles and they all give the same problem.