[asterisk-users] "automatic call bridging when destination is available" feature

[Olivier]

2009/3/14 Vieri <rentorbuy at yahoo.com>

>
> Hi,
>
> I'd like to implement the following:
>
> Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent
> to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS.
> Please try again later or dial $CODE now to notify you as soon as $EXTN is
> available.".
>
> So the "notification" part is what I'm trying to figure out.
>
> The extensions are SIP (but not all have displays) and I would like to do
> more than just  a sendtext() when the hint of SIP/102 changes. Besides, the
> SIP protocol doesn't seem to allow me to query the "real" availability of an
> extension


this is not always true as some SIP hardphones can be set to centralize DND
status on a server (with so-called Starcodes features) and in this case you
don't have to call the hardphone to query it's status,

you also can make use of SIPPEER(curcalls)




> unless I invite it and ring it (OPTIONS doesn't tell me if DND is set on
> the "client's side" - especially on softphones).


I don't know if DND is widely implemented in softphones as users might be
tempted to simply turn softphone off

>
>
> So what I would prefer to do is a two-step call bridging of 102 (dst) with
> 101 (src):
>
> 1) periodically check via events (until globally-set timeout) via
> hints/ExtensionState that both dst and src are "available"/not in use.


What would you say if before hanging up, extension 102 would just check if a
"call-back-later" request have been left (by extension 101 trough IVR) and
use call files to generate requested calls ?
This would avoid events handling.



>
>
> 2) when 1) is true then "originate" two simultaneous calls to both dst and
> src (or maybe just one call to dst). Upon answered, the originated calls
> should say something like "Automatic call bridging service: please wait
> while we connect $SRC to $DST". At this point extensions src and dst should
> be able to talk.
>
> I was thinking of making a call file just for extension dst and send it to
> a context which Dial()'s extension src. It could happen that extension src
> suddenly switches to "uanvailable" or "DND on the client's side" or "busy"
> right when the context the dst extension is sent to (via call file) tries to
> Dial(SIP/src). In that case, if DIALSTATUS is anything but ANSWERED then the
> "Automatic call bridging service" would have to say to extension dst
> something like "Sorry, failed to connect. Will try again later".
>
> Anyway, instead of starting to implement my custom solution, I would like
> to know if such a "service" already exists or if someone could share their
> thoughts. I know that some legacy PBX already have this but I don't know how
> they call it. Usually, when a caller finds a busy destination, they can dial
> *6 and the PBX will take care of bridging the two extensions as soon as both
> are "free".
>
> Can Asterisk already do this? Or is there a simple way of implementing it?


just a thought :

-

>
>
> Thanks,
>
> Vieri
>
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090314/faba5dd8/attachment.htm