Modifying sip.conf [message #147994] |
Mon, 20 April 2020 15:29  |
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Eberhard
Messages: 93 Registered: October 2008 Location: Bochum, Germany
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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 
Many thanks, Cheers
Eberhard
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Re: Modifying sip.conf [message #148331 is a reply to message #148281] |
Sat, 20 June 2020 21:54  |
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Eberhard
Messages: 93 Registered: October 2008 Location: Bochum, Germany
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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".
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