Call drops after 30 or 40 seconds while using outbound and inbound SIP with Dinstar gateway

I searched everywhere on the internet and I didn’t find any solution to fix this ugly issue

My Nat settings are all local I don’t use anything over the internet so the firewall of the router is not the problem and I already disabled the firewall in FreePBX settings

So please if anyone knows the real fix to this problem help me to wake up from this bad dream thank you :slight_smile:

The screenshot shows no SIP traffic. If your gateway trunk is using chan_sip, use
sip set debug on
to enable SIP tracing on chan_sip.

Or, the time range posted did not include the failing call.

After enabling pjsip logger or sip debug, make a failing call and paste the Asterisk log for the call (which will include the SIP trace) at As you are too new to post links, just post the last eight hex characters of the link.

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Last eight hex of the link ==> d17b6395 is the FreePBX server is the DInstar gateway that I use to make the calls , and is the IP-phone’s I used to make the calls

The number that I called is 0915555336 and the call dropped after 33 seconds I think

There are no calls attempted in that log (no INVITE requests).

So I need to repeat the call again and check the logs until I get an INVITE request ?

If the call connected, Asterisk must have sent an INVITE to the gateway, so if it’s not there, either your logging was not set up correctly (note that both pjsip logger and sip debug are turned off by Apply Config and you would need to reissue them), or you were not looking at the proper time range.

Check this log maybe you will find the INVITE request

DESTINATION: 0944423847

Call duration 40 seconds and dropped on 43 seconds

Sorry, but the only occurrences of 0944423847 in your log are on lines 421 and 437, related to a BYE request after the call had ended. Please paste a more complete log.

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