[Serusers] Adding services - help needed

[Vitaly Nikolaev]

See inline

 

________________________________

From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
Behalf Of ser
Sent: Wednesday, July 21, 2004 12:07 PM
To: serusers at lists.iptel.org
Subject: [Serusers] Adding services - help needed

 

I need help choosing how to implement different services with SER. I was
thinking to use Asterisk for application services, but i don't know how
the integration could be done.

Here is a list of services. Could you give advice of what would be the
better solution to get this ?

- Let the end user hide his calling number. Is it possible with SER to
choose to hide the calling number and stay anonymous. 

            You can do it with SER but you will need some development,
but that could be done on sip UA. It kind of standard feature

- Let the end user choose to reject anonymous incoming call. (he could
active/deactivate this function via a keypad combination)

            Use exec function and execute external program for each
invite and check if destination user has enabled anon call rejection and
if yes forward call to somewhere (to asterisk on extension with playing
rejection prompt)

- Double call with signaling of incoming second call while already on
communication.

            That called 3 way calling, SER can not help you, but it is
in all UA I know, cisco, Sipura, innomedia, etc.. you do not need to do
anything to make it working on ser

-         Call transfert : let a user transfer a call to another user
with a keypad combination.

 

-         Call transfer is ONLY UA device feature

For each case, what would be realist ?

Yea.. u can do all this stuff, I use asterisk for promts and menu.. like
user dial *68 (call forwarding always) and it goes tyo asterisk ans
asterisk ask "please enter phone number" user enter phone and asterisk
say thank you and same number to mysql.. 

 

Next when user get call from somebody, routing script check if user has
call forwarding enable and if tes tell SER where to route the call

 

 

Note that i don't want to use asterisk as the main SIP router. I want to
keep SER at front for routing logic

 

thanks.

 

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