GFI Software

Welcome to the GFI Software community forum! For support please open a ticket from https://support.gfi.com.

Home » GFI User Forums » Kerio Operator » Modifying sip.conf
Modifying sip.conf [message #147994] Mon, 20 April 2020 15:29 Go to next message
Eberhard is currently offline  Eberhard
Messages: 93
Registered: October 2008
Location: Bochum, Germany
Hi all,
has anyone a working procedure for modifying the sip.conf file?
I found some bugs in call transfers, e.g. no audio when an incoming call is routed to another outgoing line.
To fix it, or at least give it some testing, I have to modify the Asterisk sip.conf.
The problem is that some backup mechanism in Operator dismisses all changes after the next boot Surprised
Many thanks, Cheers
Eberhard
Re: Modifying sip.conf [message #148281 is a reply to message #147994] Mon, 15 June 2020 16:48 Go to previous messageGo to next message
juankax is currently offline  juankax
Messages: 7
Registered: June 2020
Hello Eberhard,
put test with double asterisk
Re: Modifying sip.conf [message #148331 is a reply to message #148281] Sat, 20 June 2020 21:54 Go to previous message
Eberhard is currently offline  Eberhard
Messages: 93
Registered: October 2008
Location: Bochum, Germany
Many thanks! But the main problem is that you have to use configuration hooks to ensure that the changes are permanent. But this post can be closed, the purpose of modifying the sip.conf was to test what the problem is that calls from extern going through call diversion have no audio. I found the solution in the extensions.conf.

I will write a new post for this as "solved".
Previous Topic: call routing error
Next Topic: blind transfer callback if busy !!
Goto Forum:
  


Current Time: Fri Jun 02 00:57:50 CEST 2023

Total time taken to generate the page: 0.05973 seconds