Ringback tone issue

Asterisk Asterisk SIP
1

Hello,

I’m currently observing the following situation:

  • Alice, with her mobile phone, can successfully call Cory, Daniel or anyone.
  • Bob, with his deskphone, can successfully call anyone but Cory nor Daniel, using any SIP trunk among those provided by two different ITSPs.
  • Cory and Daniel are both employees working in two different companies.
  • Bob deskphone is connected to a Bicom system.
  • This Bicom system is connected to an Asterisk box.
  • This Asterisk box is connected to two different SIP trunks.
  • During unsuccessful call setup, Bob doesn’t hear any ringback tone.
  • Alice and Bod are located in France.
  • Cory and Daniel are located in the USA.

When capturing unsuccessful calls, I see that:

  • as soon as my Asterisk box receives a 183 Session Progress from ITSP, it sends RTP to the other party,
  • SDP data looks OK,
  • no error is replied to outbound RTP,
  • I can’t see any RTP coming in from my ITSP.

  1. Is correct to think that in a protocol-compliant SIP, pure VoIP, call, the first 180 Ringing message is sent by callee’s phone when it’s starting to ring ? In other words, ITSP equipment should not send 180 Ringing messages to upstream if they didn’t receive any from downstream ?

  2. What are the possible causes of the issue above ?

Best regards

2

If that is the case, there are rather a lot of misconfigured Asterisk systems!

Also, if you end up on an analogue line, the network initiates the ringing, not the phone.

There is nothing that I can see wrong with the use of 183. 183 doesn’t have to be associated with early media.

3

I’m not sure I understood what you meant about “misconfigured Asterisk systems”. Do you mean intermediate systems MAY send 180 or do think they should not send it at all (only propagate those) ?

4

I mean many people code RInging() into the dialplans, before calling Dial, or use the r option on Dial.

5

OK, then !
I very often use r option in Dial statements, coupled with progress_inband settting, as a way to force opening of RTP ports (in case both call parties are in the WAN side of the firewall).

Up to now, I never used Ringing funtion at all.
At the moment, I still rate Ringing as a hack as, IMHO, you shouldn’t use it, if Early Media is used.

Anyway, back to my original question n°2, I was thinking of the following possible causes:

  • a misconfigured firewall or IPBX at callee’s end
  • an imbalanced route between callee’s ITSP and upstream provider
  • an incorrectly configured call forwarding setting (that would only touch the dialed DID)

I fail to see many thing from caller’s end as this issue only touches a couple of (international) DID and the difference between domestic and international calls, for instance occurs several hops downstream.

Maybe a misconfigured CALLERID ?

Any help on this would be much appreciated.