Feature request: redirect incoming call.

[laurent.debacker at advalvas.be]

>
> laurent.debacker at advalvas.be wrote:
> ontent-Type: text/plain;charset=iso-8859-1
>>
>> Hi,
>>
>> When a wall is incoming, minisip allows to either accept or reject the
>> call.
>>
>> I am looking for a feature allowing the subscriber, when an incoming
>> call
>> hits minisip, to have a third option, in addition to accept or reject
>> the
>> call.
>>
>> The third option would be to enter a new destination, and press a
>> "redirect" button.
>>
>> Would this be feasible? Is it in the roadmap? If not, can you envisage
>> it?
>>
>> The reason for this is the following.
>> We have an office and many remote workers. When an employee comes to the
>> office, he can be sitting at different desks (business centre).
>> Every morning, the user has to register his phone extension to the phone
>> located at the place he sits. Painful procedure, and people forget it.
>>
>> Now, the idea is to replace the PABX with Asterisk or SipX. We install
>> minisip on all PCs.
>>
>> If the worker works from home, then, incoming calls reach minisip.
>> If the worker is in the office, the incoming call reaches minisip. The
>> user can either take the call on minisip, or, redirect the call, in real
>> time, to the phone aside him, using the feature suggested above.
>>
>> Thanks for your help.
>>
>> Best regards,
>>
>> Laurent.
>
> Laurent,
>
> I know some things about SIP, but I don't speak for the minisip
> project or know what's in their roadmap.
>
> Short answer:
> I suggest looking more closely at Asterix to see if it already
> supports your need without changes in minisip.  It's reported
> to be very full featured.
>
> Longer answer:
> Is the "phone aside him" a SIP phone??
>
> In an all SIP environment, the functionality you're talking about
> is usually accomplished in the proxy or B2BUA (Asterix or SipX).
> Basically users could register multiple hard and soft clients
> and, possibly with the help of presence indications, the proxy
> would ring one or more than one phone, sequentially or concurrently.
> (Depending on the proxy's capability and configuration.)
>
> Asterix may be able to do the same, even with a mixture of SIP
> and POTS phones.  I haven't used it, but I wouldn't be surprised
> if it had a web interface to configure "where people are now".
> The user would have to remember to "register" their hard phone,
> but the first call to their minisip would remind them.
>
> While there are SIP protocol elements that could support
> "redirect from the client", you'd have to verify that they're
> supported on your chosen proxy and confirm interoperability.
> And it wouldn't be any easier to use than the (theorized)
> Asterix interface.  The only advantage is that it could get
> that first call to the hard phone without the user being
> proactive.  But who knows, Asterix might even have an
> intercept capability.
>
> -troy
> _______________________________________________
> Minisip-users mailing list
> Minisip-users at minisip.org
> http://lists.minisip.org/mailman/listinfo/minisip-users
>
Thanks Troy,

The idea, though, is to do the redirection at user agent level: when your
softphone (minisip) is ringing, you enter a new destination, and press
"redirect". This sends the SIP INVITE further to another user agent,
actually, the SIP "hardphone" located aside the user, for more comfort (if
the user doesn't have his headset ready).

The option could be enabled/disabled in the options of MiniSIP.

The hardphone is actually not assigned to any individual, but just there
for the person who is currently sitting there (remember, in a business
centre, you'll never get 2 times the same seat), to pass calls, or to
receive calls (after redirecting from softphone which is the topic of my
request).

Redirection is currently possible during the call (using some DTMF
sequences intercepted by Asterisk), but not prior to answering the call,
in the ringing phase (At least from what I know).

The idea here is to add the capability to redirect an incoming call during
ringing phase, to a given destination.

I got another suggestion to have a module interfacing with Asterisk CPL,
but, from an end user perspective, it still makes more sense to have it
implemented on the user agent.

Best regards,

Laurent.


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