SIP inbound call to my freePBX 15 through pjpsip trunk drops after 33 seconds

Look at the response starting at line 31.

Asterisk is telling the calling server to send the audio to (presumably the public IP address of Asterisk), but is telling the server to send the ACK to, which should also be the public IP address of Asterisk, but is in fact in Canada! Do you know where this came from?

Check that in Asterisk SIP settings, on the General tab, that External Address and Local Networks are properly set. Also, on the pjsip tab, External IP address and Local network should normally be blank (though it’s ok if properly set). Restart (not just reload) Asterisk and retest.

If you still have trouble, search the /etc/asterisk directory for and report where, if anywhere, that bogus address appears.

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Oops, this seems to be the trouble:

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@Stewart1 wow Thank you so much, im about to give up - setting it now and restarting the whole system
your a great Blessing, Thank you so much again. All are working now

its just a find strange all local extensions can be dialed but the ring group can’t

@Stewart1 hi there, if i may ask why is it i can dial my loca extension on my freePBX on the remote PBX but i cant dial using the ring group extensions, any idea somehow.? it just say Temporarily Unavailable

Sorry, no ESP. Log, please.

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Hi guys, i have strange problem with this existing setup please anyone can help, i got this problem that the SIP endpoint cant dial in to my freeppBX and as per checking i get this logs

Endpoint: SFDC Unavailable 0 of inf
Aor: SFDC 0
Contact: SFDC/ f6375ee0a3 Unavail nan
Transport: udp 3 96
Identify: SFDC/SFDC

anyone have idea, this was working yesterday just suddenly today i get this problem that calls are not pushing through

*posts something that is not a log*

So here is what YOU need to do before coming back here and asking again. It worked yesterday and stopped working sometime between then and now. Go through the actual log (/var/log/asterisk/full) and find where the trunk was working, then follow along until you see an error where things stop working. Research the error using the FreePBX wiki, the Asterisk wiki, and search engines. I for one will not respond any more to “anyone have idea” messages when they contain nothing useful even for speculation.

Just a guess, your public IP address changed, or your NAT settings got corrupted.

In Asterisk SIP settings, confirm that External Address and Local Networks are correctly set. If you change these, restart (not just reload) Asterisk.


Actually they are two trunks from different SIP Endpoints, i setup on trunk both two are working receiving calls but this morning the Second trunk suddenly stop accepting calls but the first one is still active with calls, so im thinking the first trunk should with affected also.?

the IP address is not change at all, all are on the same settings so far we did not change anything

Which is it? In either case, paste the relevant section of the Asterisk log for a failing call at and post the link.

the 33 seconds is already resolved with your help before all are working it just the second trunk today suddenly got problem trying to check the logs now

i see this logs on /var/log/asterisk/full


[2020-04-25 03:19:08] VERBOSE[15094] res_pjsip/pjsip_configuration.c: Endpoint SFDC is now Reachable
[2020-04-25 03:19:08] VERBOSE[15094] res_pjsip/pjsip_options.c: Contact SFDC/sip: is now Reachable. RTT: 1616.709 msec
[2020-04-25 03:21:09] VERBOSE[26908] res_pjsip/pjsip_configuration.c: Endpoint SFDC is now Unreachable
[2020-04-25 03:21:09] VERBOSE[26908] res_pjsip/pjsip_options.c: Contact SFDC/sip: is now Unreachable. RTT: 0.000 msec


hi there did you see the logs somehow.? Thanks

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