No SIP Re-INVITE when putting on hold through ARI interface

Asterisk Asterisk APIs
1

Hi everyone,

I’m using ARI HTTP calls to put a channel on hold.
Oddly, when asterisk serves my request, I don’t see any SIP message traveling from asterisk to the endpoint I put on hold.

What I’m expecting when I put on hold a channel, instead?
A new SDP offer shall be sent with a re-INVITE request that contains the “inactive” or the “sendonly” attribute depending on whether the remote side should generate a tone or not.

How can I get this behavior when using ARI? Is there a configuration file to change?

Thanks,
Daniele

2

This is only supported on the PJSIP channel driver[1]. Normally Asterisk generates the hold music itself and the hold is not passed through.This is what most individuals expect and want to happen. It is also only supported as of Asterisk 15.

[1] https://github.com/asterisk/asterisk/blob/15/configs/samples/pjsip.conf.sample#L654

3

Thanks, Joshua.

So, if I try with:

  • Asterisk 15
  • PJSIP channel driver
  • MOH disabled

I should see the re-invite SIP message, right?

4

I believe so, I haven’t tested it via ARI explicitly. It may literally mean “play hold music to channel” versus “place caller on hold” instead which would be an ARIism and would stem from the fact that it is what most people would expect/want to happen like I said before.

5

Great. I’ll try and let you know.