One way audio

First I searched the forms to see if this has happened before but couldn’t find anyone in my same situation.

Moving to a new PBX -
Finally upgrading from Elastix 2.5. Sadly I know just enough to be dangerous but not enough to fix anything major. Server is hosted at digital ocean, 3 remote sites connecting all using Ubiquiti equipment / Security Gateways.

When I switched the phones over to the new server everything worked great, after a day callers could not here us but we could head them. If I restart the server the problem is fixed and the callers can hear us again. Very strange - had no issues before with my setup on the old Elastix server.

Current sip settings, pjsip extensions (no idea i guess this is new?)



Thanks for the help guys.

PS if I need to do a packet capture or something you guys will need to help me out on how to do it. :slight_smile:

In Asterisk SIP settings, make sure that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config, you must restart Asterisk.

If that’s not it, you’ll probably have to wait for it to fail again before you can troubleshoot.

Given that your PBX is (I presume) on a static public IPv4 address with no firewall other than the FreePBX firewall, it’s hard to see why you would have trouble on the trunk side. If there is a hardware firewall at DO, provide details.

Once the failure occurs, determine the simplest thing that fails. Does *43 (echo test) work correctly? If so, do calls between extensions at the same site work correctly? If not, do some sites work and some not? If all ok, do calls between extensions at different sites work ok?

If only external calls are affected, is the problem only on incoming? If incoming calls are routed to an IVR, queue, etc., does the caller hear the announcements (but then not hear the agent)?

To get SIP traces included in the Asterisk log, at the Asterisk command prompt type
pjsip set logger on
sip set debug on
Note that these get wiped out on restart / reload / Apply Config, so reissue them after making such changes.

Then, make a failing call, paste the Asterisk log for the call at and post the link here.

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