Rtpmap: telephone-event

Hello! Ouir Bridge provider made a change and decided that we must send the telephone-event payload with our outgoing requests… does anyone know how we can do this in 15.3? :slight_smile:

Configure the appropriate SIP channel driver (chan_sip or chan_pjsip) with the appropriate DTMF mode (rfc2833 or rfc4733) and it will be placed in the SDP.

Thanks for the reply, here’s my current pjsip.conf file. Could you advise on what I’m doing wrong?


; TLS options


; snipped for security


I assume you just snipped a ton of content out of that, but dtmf_mode=rfc4733 will place it into the INVITE going out. If you did not snip a ton out of the endpoint, then it is incomplete and will do nothing.

The only thing I snipped out was the ACL list which contains a bunch of IP addresses that I can’t disseminate.

Could you point me in the direction of what I’m missing? I inherited this project from a developer that left the company in a hurry and, uh, well here I am.

The wiki[1] has examples for various scenarios, but without understanding of how the system is already configured and used it’s impossible to say precisely where to change things.

[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip

We’re using Asterisk as a middle-man to interface between our web client and various bridging providers. We send a bunch of channel originate commands via AMI and monitor asterisk’s output to make sure we’re still connected, and that’s it.

I’ll continue to dig through everything. Thanks for taking the time to help!

You should also use “pjsip set logger on” to confirm it’s not already there.

Thanks, I’ve already confirmed it’s not there, which led me down this rabbit hole.