asterisk users - Aug 2009 - Anyone actively using RLT for mobile phone forwarding?

Hello,

We currently have a scenario where a large percentage of inbound
calls on a telco PRI are intended for professors who aren't currently
in their offices/at their desks.

My question is, is anyone actively using the Asterisk "RLT" (Release
Link Trunking) feature to bounce these sorts of calls back to the
telco? The idea being to forward the call to their mobile phone without
tying up two of the PRI channels (one inbound and one outbound)
for the duration of the call.

If so, any caveats pertaining to the combination of RLT and Asterisk
that I should be aware of before attempting to build such a system?

Thanks,
Brian

This is a "hack" solution;  If the person isn't there, just hang
up the call
and the telco will handle???

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian Thompson
Sent: Tuesday, August 04, 2009 9:36 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Anyone actively using RLT for mobile
phoneforwarding?


Hello,

We currently have a scenario where a large percentage of inbound
calls on a telco PRI are intended for professors who aren't currently
in their offices/at their desks.

My question is, is anyone actively using the Asterisk "RLT" (Release
Link Trunking) feature to bounce these sorts of calls back to the
telco? The idea being to forward the call to their mobile phone without
tying up two of the PRI channels (one inbound and one outbound)
for the duration of the call.

If so, any caveats pertaining to the combination of RLT and Asterisk
that I should be aware of before attempting to build such a system?

Thanks,
Brian



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On 8/4/09, Brian Thompson <brian at eng.wayne.edu>
wrote:
>
>  Hello,
>
>  We currently have a scenario where a large percentage of inbound
>  calls on a telco PRI are intended for professors who aren't currently
>  in their offices/at their desks.
>
>  My question is, is anyone actively using the Asterisk "RLT"
(Release
>  Link Trunking) feature to bounce these sorts of calls back to the
>  telco? The idea being to forward the call to their mobile phone without
>  tying up two of the PRI channels (one inbound and one outbound)
>  for the duration of the call.
>
>  If so, any caveats pertaining to the combination of RLT and Asterisk
>  that I should be aware of before attempting to build such a system?
>
>  Thanks,
>  Brian
Hello,

We have set up several systems using TBCT(Two B-Channel Transfer),
which is the NI2 protocol version of RLT(which is for DMS100 only). As
long as the carrier supports it(which is always the biggest problem in
our experience), the actual functionality works great with Asterisk.
Some things to keep in mind:

- Your carrier will bill you for the time on the off-circuit legs of
the call, and will send you a PRI message when the call hangs up.
Asterisk throws up a warning for this event and ignores it(not logged
in any way).
- With RLT(on DMS100) you can only do RLT with calls that come in to
your circuit, whereas TBCT on NI2 can work for inbound or outbound
initiated calls.

Setup is fairly easy for these, just set zapata.conf
facilityenable=yes and transfer=yes and upon a native bridge on the
same trunk group the calls will be released to the carrier and free up
those two lines.


MATT---

On Tue, 2009-08-04 at 10:36 -0400, Brian Thompson wrote:
> My question is, is anyone actively using the Asterisk "RLT"
(Release
> Link Trunking) feature to bounce these sorts of calls back to the
> telco?
I know several people that are using Two B-Channel Transfer, which is
very similar to RLT.
> If so, any caveats pertaining to the combination of RLT and Asterisk
> that I should be aware of before attempting to build such a system?
The only caveat that stands out in my mind is that your CDR records are
only going to reflect the portion of the call up to the point that the
transfer happens... Asterisk doesn't currently do anything with the
facility message coming back from the telco when the call ends.


-- 
Jared Smith
Training Manager
Digium, Inc.

On Tue, 4 Aug 2009, Mr. Rodriguez wrote:
> I am about to initiate to prove asterisk, but that I need a Server 
> linux, that requirements recommend to me that it has my Server? we will 
> use, it for recording of calls. and reports of calls
Please hire a consultant who knows enough about project management in 
general and telephony in specific to research your needs in sufficient 
detail to ask meaningful questions.

Without any details, the correct answer to your question is "Any computer 
that can run Asterisk on Linux."

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

It depends about your traffic.
 But myabe and I guess Core 2Duo 4Gb ram , Sata 160Gb
+_
 




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________________________________
> From: clubtorreon at hotmail.com
> To: asterisk-users at lists.digium.com
> Date: Tue, 4 Aug 2009 15:37:50 +0000
> Subject: [asterisk-users] Server linux requirements
>
>
>
>
>
>
>
>
> Hello to all.
>
> I am about to initiate to prove asterisk, but that I need a Server linux,
that requirements recommend to me that it has my Server?
>
> we will use, it for recording of calls. and reports of calls
>
>
>
> saludos
>
>
___________________________________________________________________________________
>
> Carlos Rodriguez
>
> ________________________________
> ?Quieres un regalo de cumplea?os? Messenger te lo da
_________________________________________________________________