[OpenSIPS-Users] Various newbie questions

[Iñaki Baz Castillo]

El Domingo, 26 de Abril de 2009, Yehavi Bourvine escribió:

>    - *Presence*: I have here some hole in my knowledge: If I understand
>    correctly, OpenSIPS has a simple presence/dialog-state mechanism
> built-in, right? Then, why do I need an external one like OpenXCAP? Just to
> have connectivity with other presence packages?

Hi, obviously you haven't investigated about what XCAP means, you should do it 
before. Read:
  http://openxcap.org/


>    - *BLF and call pickup on Polycom phones*: This is the most hurting
>    issue, and I must solve it... Most of our users are using Polycom
> phones, and they want a BLF key which shows the other party status, and
> allow direct pickup if his/her extension is ringing. The problem is
>    that Polycom subscribes for presence and not for dialog state. Here are
> my related questions:
>     - When OpenSIPS receives the subscribe message, is there some simple
>       tool/function which allows me to modify it or create a new
> subscribe message with the relevant information?

This doesn't make sense at all. If a phone is subscribing for "presence" it 
has nothing to do with "dialog" events, nothing. Again, you should read about 
what "presence" events and "dialog" events mean.



>       - With Asterisk I program the BLF keys to send something like *7xxxxx
>       where xxxx is the destination extension. This way I can catch it at
> the dialplan and act according to the dialled extension state: If it is
> free, it
>       just dials it; if it is ringing then a pickup function is
> called, an if it
>       is busy then it works according to the busy policy (busy or
> voicemail). The
>       questions:
>        - How can I find the state of the destination extension?

This is Asterisk related.

>          - How can I pickup its call?

This is Asterisk related.

>          - *voicemail and fax*: My current plan is to use Asterisk as the
>    voicemail and media server. The main drawback is the lack of fax support
> on Asterisk voicemail. Is there some other voicemail application that does
> fax termination?

Is it an OpenSIPS related question?



Please, if you have questions about Asterisk don't ask them here.

Regards.


-- 
Iñaki Baz Castillo <ibc at aliax.net>