I am not a techie but have managed to put together our asterisk server using Freepbx as our GUI.
This morning our phones were working fine and have been since I set this up in Jan 2014 but we were having issues with connecting to a few websites, which was due to DNS error. The DNS on the router was changed and rebooted. At the same time, the firmware was updated on the router (TP Link 150M Wireless Lite N Router).
Tried making calls on our Cisco phones but only got “All circuits are busy” error message (I am based in UK). I have been at this for hours and can’t solve it. I really would appreciate some pointers.
This is the ending part of the error log
redacted logs by moderator
Forgot to mention, I also updated a few modules on Freepbx as well at the same time. Cant remember what they were though.
This is a network side error. They re likely experiencing an outage.
Hi James, I did think of that but our internet and phones are on the same network. Internet all works fine, internal calls all work fine. Just no incoming or outgoing calls.
From this portion of the log I think your PRI is down:
redacted logs per user request
Can you please give some information how to fix it. As I said initially, not much of a techie!
Call your provider tell them calls are being rejected with cause code 38
Thank you James. I shall call them on the morrow as they are closed now. I will provide an update after the call.
Hi, I have spoken to our provider, BT and they have confirmed everything is working fine on their end. We are using a TP Link TL-WR740N router. They did state to open ports on the router which we have but no difference. Thing is we never had ports open previously and everything was working fine.
Again to reiterate, all I did was change the DNS settings over from Primary to Secondary and vice versa, as some web pages were not working. Next I updated some modules on Freepbx and the phones stopped working.
This is the full error log
redacted logs per user request…moderator
I have just realised one of the modules that was upgraded was Dahdi Config module
This message has popped up
WARNING chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
There were two issues. One was the alaw codec. That was activated and once that was cancelled, I got an error message “this line has been placed out of service”
A quick call to our provider fixed the issue as they held some sort of hold on both incoming and outgoing.
Thank you James!!! You are a superstar!!