GlobalVariable = SIPDOMAIN wrong address causing Incoming calls to drop after 30 seconds

(Lewis Irving) #1

We changed NBN providers and our ip address changed from to

We are using the same SIP provider

Freepbx registers fine on the new ip but incoming phone calls disconnect after approx 30 seconds.

From the logfiles every incoming phone call has the line below

[2315] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘’

This is wrong but I do not know where FreePBX gets this info from.

I looked in the web admin and have set the external address in sip settings to the correct setting but it keeps resetting the global Variable as above on every phone call.

We cannot receive any phone call.

Outgoing calls work fine…


Please confirm that after Submit and Apply Config, you restarted Asterisk e.g. with
fwconsole restart or rebooting the entire server.
Also, confirm that you do not have External IP Address set on the pjsip tab, nor have externaddr as a custom sip.conf entry.

If you still have trouble, paste a log of a failing incoming call at and post the link here.

(Lewis Irving) #3

I can confirm that I Submitted changes and then applied them.
I did fwconsole restart. I also rebooted the freepbx box.
I was wrong about the ip address being the cause though as SIPDOMAIN now shows the correct IP
I put the log file of a failed call in
I really am at a loss as to what causes the incoming calls to drop after 30 seconds

(Lewis Irving) #4

I was wrong about the cause of this issue. After several reboots and the passing of several hour the SIPDOMAIN variable started showing the correct address. However the issue persists. I do have several installs of freepbx only two are on PJSIP trunks the others are on IAX2 but my provider FAKTORTEL are not allowing new accounts to use IAX2…

Is IAX2 more stable or have I just been lucky? If it is more stable can you recommend a provider in Australia that supplies IAX2 trunks?




In your log, the AGI took ~14 seconds to execute, definitely not normal. I don’t know why and we can investigate that further, but I’m surprised that after the four extensions started ringing, none of them were answered within 15 seconds. Is that correct?

Do these disconnects happen even if the phone is quickly answered? If so, does audio pass in both directions up until the call is dropped?

A new log including a SIP trace would be useful. At the Asterisk command prompt, type
pjsip set logger on
and make the test call in. If the drops include answered calls, answer one of the extensions quickly and keep the call up until is has dropped.

(Lewis Irving) #6

So I am not anywhere near the premises and it is a Pharmacist store (drugstore) so I can’t test during the day (hence the unanswered until the line drops) but I have been experimenting using the inbound route to both ring groups and Queues.
However when the Pharmacist called he said that he was talking to people and the line dropped, so yes it does happen when the line is answered . I did the sip log and posted it . the hangup occurs at 2020-11-26 7:56:17

(system) closed #7

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