No Ringing for the person who is calling my DID


I have a FreePBX system with a PRI card on E1 line. I have multiple inbound routes (DID) configured within FreePBX. If someone calls on those DID numbers from outside, the extension/VOIP phones ring, but the callers are unable to hear the ring. Once the extension phone is picked up, the calls continue normally. Calls between internal extensions do not have this problem.

PBX Version: 15.0.21
PBX Distro: 12.7.8-2202-1.sng7
Asterisk Version: 16.17.0

Asterisk Log file:

I found a similar topic on the forums, , but it’s a little old and I could not find a Dial Application to set r option.

What signalling protocol is used on the E1 interface?

Signaling: PRI - CPE
Framing/Coding: CCS/HDB3
Switchtype: EuroISDN

ISDN is the key point here. ISDN has an Alerting message, which indicates the phone is being rung, but I’m not sure how this relates to early media presentation of the actual tone, and whether FreePBX causes a Progress message to be sent to enable early media, or whether it relies on the network to provide the ring back tone. (I suspect this is configurable, and it might also be influenced by what is downstream.)

It should be possible to obtain ISDN protocol traces:

Thanks for helping me out here @david55 .

The url you have pointed to speaks about some log file. Is this accessible only through cli? Reading the log files through the GUI does not provide me any such info, so either I am looking at the wrong log file(full), or it’s accessible only through cli (please excuse my basic questions as I am not well versed with FreePBX).

I’ve not debugged ISDN on Asterisk, so I don’t know the details of how to obtain the log file.

However, Sangoma Support appears to show how to enable it in the CLI.

Thanks @dicko & @david55

I was able to make a log of PRI messages from the links that you shared.

I have the disconnection messages too, if that is of any help

Make sure you have g711 ulaw and alaw enabled on the sip leg of the connection, if so enabled then you should see the media flow with rtp set debug on

ulaw and alaw are enabled, but couldn’t find g711 within codecs (GUI-> General SIP settings). g726 & g722 are also enabled

*law IS g711. You should only need alaw on your PRI but some providers might also support g722.

Given a proper configuration of your ISDN trunk then the bridged media streams with a TOS 5 should appear as RTP packets rtp set debug on if you set intense on your pri debug you should see more q931 details there also.

This is what I could get after setting debug on

That indicates a successful negotiation of the Q.931 DID to the Asterisk core. Now you need to investigate the bridge created to your SIP endpoint. (the rtp thingy)

I did run “set rtp debug on”. Is there anything else I need to do to get more data for debugging?

Before the SDP session is negotiated a SIP session needs to be created, and agreed on. This you can debug using your chosen channel’s protocol’s debug phraseology or generically you could use sngrep.

Given a successful connection, then you either need to ‘answer’ the call or possibly enable ‘early media’ to hear anything.

Mostly this ‘just works’ :wink:

With sip debugging also on (sip set debug on), I do get this:

Once the call is answered, it works smoothly with audio. It’s only the ring that is not audible to the external caller. Internal calls between extensions do have audible rings on both sides.

How do I enable ‘early media’ to see if that helps ?

What endpoint should answer the call to 09811337714, apparently an Iranian number, (Iran is known to be exceedingly unfriendly to VOIP calling so the carrier might well be messing with you.)

This is an Indian cell number. The end point is a voip phone within the same network as the FreePBX server

Then explore the traffic to/from that endpoint , both SIP and any resultant SDP/RTP.