similar to: Passing an argument to a macro within an Originatecommand

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 =>

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

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 having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202

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, I'm new in asterisk and I got to set up a dundi config for my work. I have 2 PBX for the test, the two PBX are in the same local network PBX A : 192.168.199.23 PBX B : 192.168.199.21 my config files : (on PBX B , the config files on PBX A looks like it) /etc/asterisk/dundi.conf [general] bind=192.168.199.21 port=4520 cachetime=5 ttl=32 autokill=yes entityid=00:30:18:4C:33:53

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})})

For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to

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 =>

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

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:

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records despite the dialplan setting it. My program generates the calls by setting Channel=Local/NUMBERTODIAL at

Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing.

Hello all. I have installed AsteriskNow 1.7.1 with all updates. I'm able to make outbound calls without any problem (the external calls are made via an analog line, and the receiver see the CID). However I'm unable to forward incoming calls to the destination I want. What happens is when I make an internal call I ear a "bye". Bellow is the log of the internal call: --

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Hello list ! This is my call to the macro : exten => 36,n,Macro(mymacro,${KNUMMER},yo) This is where {ARG1} and {ARG2} are used inside the macro : exten => s,n,Voicemail(${ARG1}@${ARG2},s) This the output on the CLI : VoiceMail("IAX2/zoiper-13958", "908001@|s") The variable ${KNUMMER} is set in the dialplan and well replaced by 908001. But {ARG2} is not replaced by

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried