similar to: cmd Dial with U option

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Hello, I'm not understanding how to use GOSUB_RESULT in U() option from Dial app (I'm using 1.6.1) My extensions.ael is : context mylocal { 2 => { Dial(SIP/7530,,U(mynotify)); HangUp(); }; 3 => { Dial(SIP/7531); HangUp(); }; }; macro mynotify () { GOSUB_RESULT=ABORT; }; I

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

We redirect calls to another pbx - which are not under our control. So, we need to distinguish when an operator answer from the time the call stays on queue. To do so, we use dial cmd with M option. M calls a macro that repeats a digit (saydigit) until someone dial it. When operator dials the correct digit we finish the macro making it to be bridged (caller and called). It almost works

One of my sites asked for a way to identify if the person they are calling on another extension is already on another call. To that end, I wrote a bit of code in the dialplan for my extensions that checks to see if the extension they are dialing has a device status that is anything other than NOT_INUSE. If the device is NOT_INUSE, then it dials the call normally. If it has a different status,

I have an asterisk box using a SIP provider and IAX2 softphones clients. Audio is low and I need to apply some gain on it. How can I configure such gains - in/out on sip and iax2 channels? Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101106/85417a17/attachment.htm

I am setting up an after-hours on-call system. Someone calls in and requests service, and while they listen to music on hold, we dial out to several people's cell phones and home phones. We don't know if they will be answered by the employee, or by voicemail or a spouse/relative/child/pet. So we play a message that says "press 1 to accept the call" and ask employees to train

I would appreciate hearing from anyone who has figured this one out. Here's the scenario: I have a context wherein I give the called party the option to dial the digit 9. If he does so, he is transferred a la this extension entry: exten => 9,1,Playback(pls-hold-while-try) exten => 9,n,Noop(Attempting to bridge to ${agentext}) exten =>

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) Here are the segments

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Matteo Brancaleoni > Sent: Monday, July 26, 2004 5:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based > PCIISDN card): Unable to create channel of type 'Zap'

Morning, we have gateways with FXO port registered as SIP endpoint in Asterisk. To be able to use this port, the gateway ask for prefix -lets say 9- then send dial tone and here the user enter the calling number. We want to cancel this step for the users so they can enter the entire number and Asterisk will deal with the gateway. Does Asterisk have a possibility to manage this? We tried with

I have built a dialplan which dial to someone with option M. Dial (SIP/1000,,M(MYMACRO)) Both parties are SIP phones. MYMACRO expects person on SIP/1000 dial 5 (using read) then exits - and doing so it bridges my phone (SIP/2000) with SIP/1000. If SIP/1000 hangs up before dial 5 - ok the call ends. if SIP/2000 hangs up before SIP/1000 dial 5 - the macro is unaware and keeps waiting SIP/1000

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

I've been searching the archives for the proper "Wait for Dial tone" command in the "Dial(Zap/g1/18005551212)" dial sting. Does anyone have an example of it's use? Arick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040906/6f8b21a6/attachment.htm

Hi, For the dial application, parameter g is described as a.. g: When the called party hangs up, exit to execute more commands in the current context. I want the following priority (or at least a priority I can jump to) to be executed anyway, it doesn't matter which party hang up. Is there a way to do so? Thank you Dov -------------- next part -------------- An HTML attachment was

In 1.0.9, if I Dial() with the M option, the specified macro executes just fine, however there is several seconds of silence - no audio transmitted to either caller or callee. After 5 or 6 seconds (in my installation) call audio is transmitted normally. Is this known behavior? Can't seem to find a reference to it. On a different topic, has anyone noticed that google searches of the type

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register => user:pass@provider/400 [inside] exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT) exten => _4X.,2,hangup -- Executing

Hi All, suppose this: Dial(SIP/<somecarrier>/<somenumber>/60/L(3600000)M(td|${EPOCH}) where 60 is the seconds to wait for the callee (the called party) to answer L(3600000) is the absolute limit of the call once it has been answered, in ms M(td|${EPOCH}) is the macro to execute when the call gets answered. ${EPOCH} contains the current unixtime. That's the macro: [macro-td]