SIP signalling issues - call ringing

I have a very weird problem, which is that in certain configurations for inbound calls, the calling party does not receive a ringing tone, nor does the phone attached to my ATA ring. However the phone on my SIP extension displays the number of the person and it is possible to answer the phone even though it is not ringing, My setup is as follows:

Provider (voip.co.uk)
--> pbxes trunk
--> ring group (1 x sip extension, 2 x classic extensions - mobile numbers)
--> ATA (Fritz!Box 7270 functioning as router)
--> DECT phone

I had this problem previously with a different provider, and I solved it by changing the order of the extensions in the ring group.

It started doing the same with voip.co.uk when I removed the second routing to a softphone within voip.co.uk configuration, and used pbxes.com to do the routing instead.

It appears that there is some sort of conflict somewhere, but I can't work it out at all! All hep gratefully appreciated!

RE: SIP signaling issues - call ringing

In general, ringing is controlled via two Informational Responses in SIP: the 180 Ringing and the 183 Session Progress.

The main difference between them, is the 180 Ringing message instructs the UA to create the dial-tone locally, whereas the 183 Session Progress contains an SDP, which allows for regional ring-back and carrier announcements as well.

Both messages originate from the local SIP Proxy, based on the incoming Invite from the far end GW or B2BUA, and are propagated to the remote UA.

So, in your case a 180 Ringing or 183 Session Progress message, should be received on your ATA from PBXes, based on the Invite of proxy.voip.co.uk, for the attached phone to ring. Is such message received by your ATA?

To eliminate a possible Ring Group issue, set the Inbound Route directly on the ATA. Then verify you get proper ringing on your phone, as well as your caller gets proper ring-back.

Keep in mind that you shouldn't expect to get proper ring-back on the Classic extensions, when they are included in the "extensions list", only when they are set as an extension in the "Destination if no answer" section.

RE: SIP signalling issues - call ringing

Well, after several days of dealing with this problem, I can say the following:

The fault is intermittent. I get no problems when I try it myself (using a mobile) but my callers are often complaining of no ring tone, and the ata does not generate a ringing on the telephone line.
However I configure ring groups, I do not have a reduction in the number of problems, even when only one SIP extension is specified in the ring group.
Routing the calls directly to my ATA from proxy.voip.co.uk does not generate issues.
Issues seem to be reduced when forking from proxy.voip.co.uk to two SIP registrations (one of them my ATA, the other pbxes.com).

In short, PBXes is the only common link at this stage although I have only now tried configuring inbound routing to an extension instead of a ring group.

If you want to take a look at the system logs for a problem call, try looking at my logs from Sep 16 17:31:05. This was a call which the caller complained that they did not receive a ringing tone. Unfortunately, my ATA does not provide detailed logging in it's web interface so I can't check the other side of the communication.

Regarding your comment on no ringback for classical extensions, do you mean the caller will never hear a ringing tone?

RE: SIP signaling issues with ring-back

Regarding my last comment: "Keep in mind that you shouldn't expect to get proper ring-back on the Classic extensions, when they are included in the "extensions list", only when they are set as an extension in the "Destination if no answer" section".

Since a Ring Group's function is to instruct a collection of extensions to ring at the same time (if the ring strategy is set to "ringall"), it assumes all the extensions will get the 180 Ringing message at approximately the same time (within a few microseconds). So a Classic extension which requires the use of a trunk found on another SIP Proxy, which in turn needs to signal a PSTN gateway found in another network, introduces additional delay in the call setup time.

In the case of a 183 Session Progress message the delay is further elongated, since the ring-back is required to arrive from the remote Class-5 switch (MSC in mobile networks), so the native ring-back can be preserved. So, if a Classic extension is part of a Ring Group instead of its' "Destination if no answer", I don't expect the caller to hear any ring-back. Almost all SIP trunk providers outside N. America use 183 Session Progress.

In order to setup a proper test environment, sent several PSTN calls to a DID which will ring to an extension. Observe the results and let us know. You might have to originate calls from a local fixed line, in addition to the calls from your mobile, since they are on different networks.

RE: SIP signalling issues - call ringing

Looking in the log I noticed that sometimes call progress goes as:
Calling <extension> => SIP response 100 => SIP response 180 => SIP response 200.

But it can also skip SIP 180 and go from SIP 100 straight to SIP 200, in which cases I there’s no ringback.
What it depends on? Why in some cases the person who dials SIP extension hears ringback, but sometimes it’s total silence while the other party answers?

RE: SIP signalling issues - call ringing

What I sometimes see is the following.

I have 2 extensions registered through a fritzbox. 2 dect phones are pluggied into the box.
I have one pstn extension which dials my mobile number
And I have 1 softphone extension on my computer.

When a call comes in the extension on my pc is the first to register the call. Than it sometimes takes at least 1 ring for the fritzbox to lett het dect phone ring but at the same time my mobile (pstn extension) rings.

so it looks like the fritzbox ata is giving me some sort of delay for whatever reason.
In my ring group pstn extension is set first than 2 fritzbox extensions than the pc softphone extension.