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.
ISDN is the key point here. ISDN has an Alerting message, which indicates the phone is being rung, but Iâm 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:
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 itâs accessible only through cli (please excuse my basic questions as I am not well versed with FreePBX).
*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.
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)
Before the SDP session is negotiated a SIP session needs to be created, and agreed on. This you can debug using your chosen channelâs protocolâs debug phraseology or generically you could use sngrep.
Given a successful connection, then you either need to âanswerâ the call or possibly enable âearly mediaâ to hear anything.
Once the call is answered, it works smoothly with audio. Itâs 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 âearly mediaâ 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.)