Unicasting audio from incoming SIP calls

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

What exactly are you dialing?

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

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

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()