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.
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 ?
What are the possible causes of the issue above ?
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.
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) ?
I mean many people code RInging() into the dialplans, before calling Dial, or use the r option on Dial.
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.
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