Unicasting audio from incoming SIP calls

Asterisk
1

Hello,

I am very new to Asterisk, and I was hoping someone here would assist me. I am using the latest GitHub version (as of June 20th.)

I am interesting in streaming the audio (via Unicast) of each incoming SIP call. I found this blog post which seems to detail how this is done, but I cannot get it working because I get a segment fault each time reaches the ‘originate’ command.

System:
Distributor ID: Ubuntu
Description: Ubuntu 16.04 LTS
Release: 16.04
Codename: xenial

Here is the blog post I’m referring to:
http://www.joshua-colp.com/broadcasting-asterisk-conferences/

And here is my dialplan on PasteBin:
fB5jfnuY

GDB backtrace of the core dump on PasteBin:
8GtH1ZGm

2

What exactly are you dialing?

3

I have Asterisk/PJSIP registered with a SIP provider, and I’m dialing the number assigned to the box with Jitsi (I only have G722 audio enabled on Jitsi)

I’m dialing the ‘15555555555’ number

4

What are you dialing for the unicast RTP driver. The exact dial string.

5

Is this what you mean?

I’ve tried two methods:

  1. Having the call go directly to streaming@livestreaming, then sending the CLI command below…

    *CLI> channel originate UnicastRTP/127.0.0.1:61001//g722 extension streaming@livestreaming

  2. Using a dialplan to initiate the originate command (see below)

    ; extensions.conf

    [general]
    static=yes
    writeprotect=no
    clearglobalvars=no

    [from-sipprovider]
    exten=>15555555555,1,Goto(auto-stream,prepare,1)

    [auto-stream]
    exten=>prepare,1,Originate(UnicastRTP/127.0.0.1:61001//g722,exten,livestreaming,streaming,1)
    exten=>prepare,n,Hangup()

    [livestreaming]
    exten=>streaming,1,ConfBridge(5555,conferences,USER,user_menu)
    exten=>streaming,n,Hangup()

6

@jcolp sorry for tagged you, I was trying to access to your tutorial for broadcasting an Asterisk conference, but looks like have been moved is there any new link ?

7

I’m afraid not, no. Not currently at least.