similar to: Call Recording using MixMonitor - close, but would like some more words of wisdom.

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hi, I have problem with runuser utility and permission to /dev/null. runuser is called from init script. When init script is run /etc/init.d/callrec start, then there are no problems with permissions, but when init script is called via service callrec start. Then runuser starts complaining about permissions. runuser: /dev/null: Permission denied What is strange is that I do not redirect to

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

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.

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

Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works. Cases that do work are as follows... Calls using both Queue() and

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

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com> wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no longer works. > > > > It only

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 =>

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

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

I'm trying to actually use the example application map in features.conf: testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app

Hello, I wonder if there is a way to build my own asterisk application (let us say apps/app_myappl.c), and to launch other existing applications from it (for example, doing an apps/app_dial.c, or others). Could someone highlight me on that? thx Pirlouwi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

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 have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

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