FreePBX | Register | Issues | Wiki | Portal | Support

Extension to Extension Follow me settings Error


(J) #1

Hello,

We have a remote employee who has had issues with calls being transferred from his primary internal extension to a second internal extension located at a different site. Whenever he works at the location where the second internal ext is the calls sometimes go straight to voicemail instead of ringing the second extension. The issue is resolved whenever I re-apply the configuration. Any thoughts as to why this occurs and how to correct the issue for good? Attached are the follow me settings.


(Dave Burgess) #2

Double check that the phone is staying registered. You should be able to look through the /var/log/asterisk/full log for the extension number to find when the phone is reachable and when it isn’t.


(J) #3

Here is the log, my apologies I’m very new to working with freepbx. The ext is 20502. Seems as if it wasn’t reachable at one point then I reset the line and it was? Not sure how to read these well.

Please advise, thank you.


(Dave Burgess) #4

You need to go back through the /var/log/asterisk/full log and find out why it’s deregistering. The message on that event should give you the hint you need.

Chances are, one of the devices (router, firewall, phone, etc.) is dropping the NATed UDP channel and the phone is just being ignored out of connection.


(J) #5

Thanks for the prompt response Dave, I really appreciate your help with this issue. This is my first exposure to these logs and i’m not sure how to fully decipher them yet. Here is the complete output for line 20502. Can you tell me what is causing this to occur and how to fix it? Thanks again.


(Dave Burgess) #6

Thing 1: Almost none of this is for extension 20502. A lot of it is for a Timecondition check for an incoming call. The rest appears to be for extension 10802.

Thing 2: The only like that appears to be about extension 20502 is the line where the phone re-registers and wakes up the phone. That sounds like either a network failure (causing the phone to go unreachable) or that your re-registration of the phone back to the server is exceeding your NAT timeout.


(J) #7

Got ya. I think I know why this is occuring thanks to your guidance. The RTP keep alive was set to 0 I set it to 30. I’m hoping this resolves the issue once and for all. What do you think?


(Jared Busch) #8

That screenshot also looks like an old version.

If you upgrade to 14 you can use PJSIP and not need a different extension for the second office. Both can be registered to the same extension.


(J) #9

Thanks for the suggestion Jared, I did not know that! In the meantime I need to apply a fix though. Looks like the same extension if having the same problem even after I changed the keepalive duration.

Here is the log showing 20502 going from reachable to unreachable. How do I fix this??

[2019-05-17 16:39:13] SECURITY[27591] res_security_log.c: SecurityEvent=“ChallengeSent”,EventTV=“1558136353-
537895”,Severity=“Informational”,Service=“SIP”,EventVersion=“1”,AccountID=“20502”,SessionID=“0x7fcec01f1008”,LocalAddress=“IPV4/UDP
[2019-05-17 16:39:13] SECURITY[27591] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“1558136353-
573325”,Severity=“Informational”,Service=“SIP”,EventVersion=“1”,AccountID=“20502”,SessionID=“0x7fcec01f1008”,LocalAddress=”
[2019-05-17 16:39:13] NOTICE[27623] chan_sip.c: Peer ‘20502’ is now Reachable. (31ms / 2000ms)
[2019-05-17 16:40:17] NOTICE[27623] chan_sip.c: Peer ‘20502’ is now UNREACHABLE! Last qualify: 31
it’s happening about a minute apart. Any suggestions? Thanks for being patient with me everyone.


(system) closed #10

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.