similar to: Goto invalid extension doesn't go to 'I' when in a macro.

Thanks a lot Jiefei, I had thought of defining a binary operator (inspired by pipes) or simply using an additional condition in the if() calls [e.g. if(foo & fn(bar)) doSomeThing; with fn(bar) returning a logical], but both are workaround that I do not find as elegant as a proper control-flow construct. Thus two questions remain: - is it possible to create a control-flow construct in pure R?

Hi Alexandre, I'm not an R expert so this is only my personal thought: I don't think you can achieve what you want exactly. A possible solution would be defining a binary operator %*%, where you can replace the asterisk with any function name you want. The function %*% is special since it has two arguments, left operand and right operand respectively. You then can call the `substitute`

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

The last Originate() option is ignored if using 'app'. It is only there for 'exten'. http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate tells all :) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce Hopkins Sent: 15 March 2011 21:36 To: asterisk-users at lists.digium.com

Hi, With Asterisk 1.8.3, I can't figure out how to pass an argument to a macro which is used within an originate command. Here is my sample dialplan to illustrate: exten => 123,1,Answer() exten => 123,n,Originate(SIP/20,app,Macro,foo,bar) exten => 123,n,NoOp(This is the NoOp after the originate command) exten => 123,n,Wait(30) exten => 123,n,Hangup() [macro-foo] exten =>

Hi, Is there some more thorough documentation of this change that has happened in 1.6? The upgrade.txt and changes.txt files mention it, but I have already seen details of this change that do not appear to be documented except in conversations on the mailing list... 1) It appears that it is no longer legal to have: [macro-contaxtA] ...stuff... [contextA] ...stuff... Is this true? Or have I

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

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]

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

I've been using the following macro to ring SIP and IAX devices for a few seconds, and then add on a cell phone if there is no answer on the SIP or IAX device. Periodic problems began a few versions ago and now the problem happens every time with 1.2.9 and 1.2.9.1. The problem is that when a call from the PRI falls through to voicemail, the call is dropped before the voicemail greeting

Howdy, I'm working on a macro that authenticates the calling extension against a list of allowed extensions but it looks like the Expression I'm attempting to send of pipe separated extensions is showing up as additional arguments to my macro. I expected to have 4 arguments to the below macro, Instead it looks like I'm actually getting 6. I'm open to suggestions to other ways

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>

in a simple dialplan like follows: [firstcontext] include => secondcontext include => thirdcontext include => fourthcontext [fourthcontext] _03X.,1,Goto(${EXTEN:2},1) _X.,1,DoSomething() _X.,2,Hangup() the Goto() for exten _03X. seems to start the search for the jump within firstcontext, thus possibly matching an exten in secondcontext or thirdcontext first before hitting the

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat => 9 exten => 200,1,Macro(stdexten,200,SIP/dzalewski) [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})})

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi all I need to pass a parameter in a thread-safe manner to the Queue pickup macro. This is to know when (and who) picked up an incoming call to a queue and log that to my back-office system with a CURL to a HTTP endpoint. However, the Queue application does not appear to allow passing of parameters to the called queue pickup macro. E. g. non-working code is: [queuetest] timeout = 60 retry =

Hello, I'm using AEL2 (in Asterisk 1.6.1.6) and I can't find a way to successfully come back into my dialplan. I've tried things like this (in features.conf) : toto => #9,peer,Goto,mylocal2,s,1 But typing #9 (from channel SIP/7275, in example bellow) I've got: -- Feature Found: toto exten: toto -- Started music on hold, class 'default', on SIP/7275-08b7fbe0