[OpenSIPS-Users] Newbie questions about OpenSIPS
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Tue Dec 9 22:34:35 CET 2008
Hello Yehavi,
Yehavi Bourvine wrote:
> Hello,
>
> Up to now I used Asterisk as our pilot phone exchange (around 20
> extensions). I am now installing and learnning OpenSIPS/Kamailio to be
> used as a front end to Asterisk in order to support much more
> extensions. I have a fe questions:
>
>
> * As far as I see Kamailio and OpenSIPS are quite interchangeable
> in the sense that I can take the config file from one of them
> and use it on the other. Am I right?
>
not really - OpenSIPS has new modules and new parameters for other
modules, so the "compatibility' is not something you should rely on.
>
> * How do I generate music on hold? I understand that OpenSIPS is
> not a media server thus it cannot generate it, but can I use
> Asterisk to generate it? If so, is there an example config for
> openSIPS and Asterisk?
>
you need to keep * on the call path in order to get such a functionality.
>
> * When an extension is busy I have some code in Asterisk to wait
> for the busy side to hangup and then generate a call to the
> original caller (here this feature is called "Call again") using
> the call files. . Is there some way to generate such a call in
> openSIPS like the "call files" in Asterisk, or will I have to
> use Asterisk for this feature?
>
I would say you still have to use * for that - CALL AGAIN is a PBX like
feature
>
> * And a different topic: When using more than one table in a MySQ:
> server, is there any difference between calling the function
> modparam("acc", "db_url" ...) for all tables at once, or can I
> call it for each one separately without having any performance
> issues?
>
you cannot use more than one table for accounting :)
Regards,
Bogdan
>
> Thanks! __Yehavi:
>
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