No ringback for German Telekom

Hi!

We can reach all destinations with ringback, just German Telekom does not trigger ringback.
This is what I tried:

  • setting “r” in Dial to outbound trunk: no change, ringback missing.
  • setting “R” in Dial to outbound trunk: wrong syntax.
  • calling “Ringing()” before Dial: very short ring, then silence.

Asterisk version is 13.18.2.

German Telekom is known for problems like this but we need to solve this on our side.

Any ideas?

Thank you.

edit:
We tried the “Ringing()”-approach again on our test installation and it worked.
The change will be applied this evening on out prod infrastructure and we will test.

edit2:
We noticed that “Ringing()” only works for some devices, not all.
There is always a short ring, some phones-models then play ringback, some do not.

Have you tried progress() before the dial command, and verify if you get ringback tone, also a SIP trace could help, as ringback tone are transmited as early media

Early media is typically supported by the use of the 183 Session In Progress response.

I did a deeper check with my colleagues. As far as I understood, German Telekom just sent 180 to our carrier which they forwarded to their SBC connecting to us. This SBC then translated 180 into 183/SDP but RTP was lacking audio. (So far this is plausible). They fixed ringback on their side and now we indeed get ringback when they passthru our call.

Seems like some phones need “inband_progress” set to yes.
I noticed that some internal calls were also missing ringback (I did not check if the destination sends 180 or 183/SDP).
Is it safe to set inband_progress to yes for all phones? CPU ressources are no problem for this setup (Xeon Server).

when inband_progress is set to yes, chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing as audio.

The only reason for me why inband progress is not usually desired, because it requires extra Asterisk resources to run a generator to generate the inband ringing. If the endpoint is generating its own, then there is no need to tell Asterisk to do it.

Also it is docummented on the following link

https://www.voip-info.org/wiki/view/Asterisk+sip+progressinband

Why do you need this “ringback” feature? I have a couple of German Telecom lines that work fine and all phones are ringing when dialed.