Newbie questions about OpenSIPS

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Newbie questions about OpenSIPS

Yehavi Bourvine
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?
  • 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?
  • 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?
  • 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?
                                             Thanks! __Yehavi:
 

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Re: Newbie questions about OpenSIPS

Bogdan-Andrei Iancu
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:
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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Re: Newbie questions about OpenSIPS

Jiri Kuthan
In reply to this post by Yehavi Bourvine
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?


unfortunately not. I recommend the SER/openser merging project to you as
there is strong effort to consolidate these differences. That's different
mailing lists though.


>     * 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 really need media server. Either SEMS or Asterisk are obvious choices,
but that's best served at other mailing lists too.

>     * 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?

you need SEMS for this.

>     * 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 can have one parameter for all of them. Something like this:
modparam("speeddial|auth_db|usrloc|acc_db|domain|uri_db|gflags|avp_db|db_ops|flexroute",
"db_url", "mysql://ser:heslo@localhost/ser")

see SER's OOB config file.

-jiri

>
>                                              Thanks! __Yehavi:
>  
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: Newbie questions about OpenSIPS

Klaus Darilion
In reply to this post by Yehavi Bourvine
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?

If you compare Kamailio 1.4 and OpenSIPS 1.4 then the config file syntax
is more or less the same.

If you are comparing trunk version (which will become 1.5) then you get
new features which differ.

regards
klaus

>     * 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?
>     * 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?
>     * 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?
>
>                                              Thanks! __Yehavi:
>  
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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