[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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