Hi Everyone,
For some reason, my system lost all audio for inbound calls. I didn’t change anything … hadn’t even logged into the pbx in months. Outbound calls work fine. Sometimes, inbound calls work, but almost all are failing.
In the network traces for failed calls, I see my end reaching out to the RTP server, but no return traffic. Calls that work, I see traffic in both directions.
My provider is Flowroute and they are telling me there is a NAT problem. They see:
Contact: sip:w.x.y.z:5160
where w.x.y.z is my internal IP address. They are telling me that the connect should show the NAT’ed external public address.
Odd thing is, on calls that work and those that fail, the contact line is the same. They tell me NAT problems can cause inconsistent audio issues. Grrrr…
Anyway, in the PBX under Settings -> Asterisk SIP Settings -> Chan SIP Settings, I have it set to Dynamic IP and for Dynamic Host I have specified the hostname of my external address. I have made no specific settings in the PJSIP tab.
Keep in mind, this is the way it’s always been set up and has worked flawlessly for months, and then suddenly it broke.
I have both SIP and PJSIP trunks so I’m a little worried about randomly changing values. If anyone can give me some specifics on where to look, that would be helpful.
It’s just so odd… calls that route to specific RTP IP’s work fine. Calls that route to others fail. I’ve just never seen this behavior before and could use some help.
Thanks to you all in advance!
FreePBX 13.0.197.28 / Asterisk 13.10.0
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