asterisk users - Mar 2009 - "automatic call bridging when destination is available" feature

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

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.

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?

Thanks,

Vieri

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
>
>
>
>
>
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