similar to: Asterisk perl AGI confusing variables

Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi, I have a dataset from ILO, originally in csv-format, that I have read into R. It is cross-sectional time-series data, so I have a bunch of variables and dummy variables that I need to extract data from for the entire time period. However, the years are separated by columns instead of rows, as is usually the case in R. This is what it looks like: > str(laborstafinMFBA)

Hi, I am trying to do a multiple regression on the dataset "Hdma", available in the Ecdat package. The data looks like this: > str(Hdma) 'data.frame': 2381 obs. of 13 variables: $ dir : num 0.221 0.265 0.372 0.32 0.36 ... $ hir : num 0.221 0.265 0.248 0.25 0.35 ... $ lvr : num 0.8 0.922 0.92 0.86 0.6 ... $ ccs : num 5 2 1 1 1 1 1 2 2 2

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

Eric, think I came across an issue with the parser in unicorn, with a request (due to 2 layers of nginx proxying) coming across with both a X_FORWARDED_PROTO and a X-Forwarded-Proto header. From the socket (in HttpRequest) - we get: X_FORWARDED_PROTO: http X-Forwarded-Proto: https which is parsed to HTTP_X_FORWARDED_PROTO"=>"http,https There was a passenger ticket that

Hi i want use the API on my asterisk 1.6, but i have a small problems : In extension, i start it : exten => _X.,3,AGI(My-Script.agi) The perl agi file are started without problems but i want get into this script a lot of variable: Type (SIP or IAX) src (from cdr) but that's don't work: use Asterisk::AGI; use lib "/var/lib/asterisk/agi-bin"; $AGI = new

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

Hi list, What possibilities are there in asterisk to send an *alphanumeric DTMF*from/to asterisk !? Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/e62e7890/attachment.htm>

Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of "hangup". I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to

My regexp skills are somewhere infinitesimally close to zero. I have never really 'gotten' them. That said, I have spent a couple hours already search for help to write a rewriterule that works on a string in the URL. In particular I want success if either of the following were provided: webmail.domain (e.g. webmail.foo.com) server/webmail (e.g. www.foo.com/webmail) And I have not

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

I tried: RewriteRule ^webmail\.|/webmail https://%{SERVER_NAME}%{REQUEST_URI} [L,R] But that does not rewrite for http://webmail.domain On 02/22/2017 06:41 PM, Robert Moskowitz wrote: > Seems I left off one point in this message. > > This is to refine these rules in my Apache server. > > RewriteCond %{SERVER_PORT} !^443$ > RewriteRule ^.*$

We have 1 queue called helpdesk and are setting up a second one called isp. The helpdesk queue is for internal support calls and isp for our ISP customer calls. Both of these queues will be directed to the same agents (helpdesk phone extensions). We want to have the separate queues for tracking purposes but the queued calls need to be ordered and answered as if there was only one queue. For

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hello everybody, I got tree UPS devices connected to a Debian server, and i am searching for a way to configure them. The server power-cord is only connected to one of the ups devices, and should monitor the other ups systems to sent email status. I got a HP R3000XR UPS with has a RS232 communication port that I have working perfectly. Then I have two APC SmartUPS 750V devices that are

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

I'm working on a Java implementation of portions of LLVM and have a question about licensing. The project won't contain any LLVM source code although I can foresee using JNI stubs that link against LLVM libraries. It does however in many cases follow LLVM APIs. From my understanding of the University of Illinois/NCSA license this means: 1. I need to include the copyright notice

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? regards, Asif