PJSIP Extensions unreachable, and then immediately reachable?

Asterisk Asterisk SIP
1

I’m having some users report are having audio issues and that they are unable to hear the person on the other end of the phone. I recently upgraded our PBX system to FreePBX v14 and I’ve changed all the extensions and trunks to PJSIP, the older PBX system was running Canh_SIP for both extensions and trunks, I’m not sure if that is the culprit? I have noticed the Asterisk CLI, that some extensions are becoming unreachable and then immediately becoming reachable. I’m not sure why, it’s not all the extensions, just some?

– Contact 7782/sip:7782@IP_ADDRESS:5060 is now Unreachable. RTT: 0.000 msec
== Endpoint 7782 is now Unreachable
– Contact 7782/sip:7782@IP_ADDRESS:5060 is now Reachable. RTT: 133.946 msec
== Endpoint 7782 is now Reachable
– Contact 7609/sip:7609@IP_ADDRESS:5060;line=08379c541225c88 is now Unreachable. RTT: 0.000 msec
== Endpoint 7609 is now Unreachable
– Contact 7782/sip:7782@IP_ADDRESS:5060 is now Unreachable. RTT: 0.000 msec
== Endpoint 7782 is now Unreachable
– Contact 7780/sip:7780@IP_ADDRESS:5060 is now Unreachable. RTT: 0.000 msec
== Endpoint 7780 is now Unreachable
– Contact 7609/sip:7609@IP_ADDRESS:5060;line=08379c541225c88 is now Reachable. RTT: 12.439 msec
== Endpoint 7609 is now Reachable
– Contact 7782/sip:7782@IP_ADDRESS:5060 is now Reachable. RTT: 78.510 msec
== Endpoint 7782 is now Reachable

FreePBX 14.0.1.20
Asterisk Version: 13.18.3

2

Unreachable occurs if we send a SIP OPTIONS request and receive no response. In your scenario this happens and then they become reachable, so it is able to answer a subsequent request. If they really were unreachable for some reason then this would also explain why they can’t hear the person.

1 Like
3

Thanks for the quick response, I really appreciate it.

So my question is now, is there a way I can resolve this? Stop them from being unreachable because its causing disruption to the users?

4

You’d want to confirm that Asterisk is sending the media using something like tcpdump, wireshark, or “rtp set debug on”. If Asterisk is sending the media then your question is outside of Asterisk and you’d need to investigate why they are unreachable.