Questions tagged [elastix]

an open source software package that joins IP PBX, email, IM, faxing and collaboration functionality.

89 questions
-1
votes
1 answer

Asterisk, force timeout delay between consecutive inbound calls

We have one problem we've been suffering of for a long long time,It's the unknown callerID received from asterisk that happens on specific situations. First we have a sip soft phone (sipml5) and on server side we…
Louay Al-osh
  • 3,177
  • 15
  • 28
-1
votes
1 answer

Asterisk (Elastix 4.0.0) Misc Destination Call Hangup Issue

i'm using Asterisk (Elastix 4.0.0) as my VoIP Server. I'm using TDM2400P card as my DAHDI trunk i set Misc Destination to outside mobile number.. calls are forwarding correctly.but when i hangup from outside my server dose not hangup the telephone…
-1
votes
1 answer

Play an audio in conference room on key press by participant in Asterisk

I have created a conference room in asterisk via confbridge.conf. Now, I want an audio file to be played to all participants in conference room when any participant press a key. I am not sure how to do that. Any help?
AmirA
  • 133
  • 2
  • 15
-1
votes
1 answer

Call Transfer, do not disconnect unless connected

I have elastix PBX, and when I transfer a call to other extension, I got "Redirect Success" even if the extension I transferred the call to not answered or busy. The case Is: A is on call with B. B transferred the call to C. B got "Redirect Success"…
Abusnake
  • 168
  • 11
-1
votes
1 answer

Build a hosted Call Center Solution/ IVR by extending FreePbx/Elastix

I m currently using Elastix as IPPBX solution. Would like to know the possibility to extend it as a hosted solution where few clients can be added to the application. Was planning to have a web application which needs to be developed so that the…
user2695433
  • 2,013
  • 4
  • 25
  • 44
-1
votes
1 answer

PFSense QoS for VOIP Central. Reserved bandwith

I was wondering if I can reserve an specific bandwith to my VOIP central using PFSense. The reason is that I have lots of users using my network and when they generate a lot of traffic I get issue with my phones. I don't think that should be…
CTala
  • 315
  • 4
  • 13
-1
votes
1 answer

Asterisk Try Another If First is Busy

I forward incoming calls to external numbers. I do this with Follow me module for each number. Sometimes I use one more external number in follow-me list, to call the second ona if the first one is busy. Bu it is calling the first one and ringing.…
Fatih K.
  • 393
  • 3
  • 6
-1
votes
2 answers

How can I set a variable = null in for loop?

I have this code in Elastix2.5 (CentOS): for variable in $(while read line; do myarray[ $index]="$line"; index=$(($index+1)); echo "$line"; done < prueba); This extract the values for each line from "prueba" file. Prueba file contents passwords…
juanute
  • 11
  • 1
  • 5
-1
votes
1 answer

Elastix Hyla FAX Says No Local Dial Tone

I have an Elastix Server configured Virtual FAX with IAX extension 123. When I am trying to send a FAX it shows "No local Dial Tone". Please anybody help me.
bujail
  • 53
  • 1
  • 11
-1
votes
1 answer

Elastix - Prioritize incoming DDI

i need a little help regarding the queues. I have a situation where i want to set an incoming number as VIP number so whenever that number calls in, it jumps to first place. For example, we have 20 calls in queue, VIP number calls in, he…
user1972670
  • 397
  • 1
  • 4
  • 15
-2
votes
1 answer

How to determine if client or agent hung up?

I have an elastix server4, but I have a big problem. I don't know how to find who hung up the call (agent or client). How do I determine if client or agent hung up?
MLazar
  • 1
-2
votes
1 answer

How can we make call through extension to extension in web using Elastix server?

I am working elastix server api and I want to use call by web(PHP)like softphone. For exmaple: I have two extenstion of elastix server - 1000, 1001 with Domain server(elastix) XXXXXXXXX and secrate key. Please suggest how can we call from 1000 to…
Pankaj Kumar
  • 94
  • 10
-2
votes
1 answer

Attended Transfer to gxw410x sip trunk Failed

I have an issue making an attended transfer to fxo gateway (grand stream gxw4108). I am using feature code (*2) to commit in call attended transfer. Call first is initiated and then transfer terminated just when the external pstn phone ring. Blind…
Ahmed Gawad
  • 135
  • 1
  • 10
-3
votes
1 answer

CentOS Elastix Two Routes In Linux

Have the next problem in my LINUX ASTERISK ELASTIX SERVER Have two interfacaces: eth0: inet addr:192.168.1.240 Bcast:192.168.1.255 Mask:255.255.255.0 eth1: inet addr:10.7.227.110 Bcast:10.7.227.111 Mask:255.255.255.252 The eth0 is the Internet…
user2001782
  • 25
  • 1
  • 5
1 2 3 4 5
6