[asterisk-users] Transfer Call to Cell Phone

[John Faubion]

>We do have full features on our lines so both lines are free once the
>transfer is complete. We also have toll calls on our lines so it would
>not be a problem, so I do not have to worry about AT&T dropping the

The issue really isn't whether you have the ability to make toll calls on
your line. The concern here is in what the regulatory agencies call "toll
bridging" which is using a system to relay a call from one local calling are
to another local calling area to avoid a toll charge. This is one of those
gray areas that can become a problem if your not careful. The problem comes
up if you have customers that can call you as a local call and you are
forwarding them on to another party that is a local call for you but would
be a toll call for the customer. This is essentially what toll bridging is
about. Now your not likely to have to worry about the legal ramifications of
this since your merely connecting the customer with an extension of your
company, namely your salesman. Where this could become a problem for you
would be in transferring the customer using the same pots line. The reason
is that AT&T is handling the transfer. When you transfer the call, it
essentially becomes a new call. The main difference is that you have
provided the called number. So the software in the Class 5 (End office)
switch, takes the number you provide and runs the call through its routing
translations (similar to the Asterisk dialing plan) and if it determines
that the destination number is outside the originators Local Area Transport
Area or LATA, then it will either drop the originator to a message that
says, "You must first dial a 0 or 1 before calling this number" or it may
deny the transfer allowing you to stay connected to the customer. Neither
one looks very professional. The only way around this would be to provide
another line or trunk to pass the call down. Now if your not in an
overlapping LATA this probably isn't an issue.


>The only way I can get it to work is by have the call on the 1st
>line then transfer it out on the 2nd line. After that is complete both
>lines are free.

Are you saying that you are able to route a call from line 1 to line 2 and
have the call transfer, thus freeing the lines or that once the call
completes the lines are freed? I've never seen the first scenario. The
second scenario is the normal behavior.


>Can you give an example of creating an extension which points to a cell
>phone. Secondly how can you have if no one answers an extension it dials
>the cell number next. That maybe answered in the example.

In extensions.conf use something like this.
[global]
SIP-PROV = "sip.urprovider.com"
; Now set the call forward numbers
CFN21 => "5555551234"  ; These are normally set in an external file

[internal]
exten => 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})

[macro-stdext];
; Standard extension macro:
;   ${ARG1} - Device(s) to ring
;   ${ARG2} - Our call forward number
exten => s,1,Dial(${ARG1},10)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}>0]?s-CFWD,1)
exten => s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
exten => s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
exten => s-CFWD,1,Dial(SIP/${ARG2}@${SIP-PROV},20)
exten => s-CFWD,2,Goto(s-NOANSWER,2)
exten => _s-.,1,Goto(s-NOANSWER,2)
exten => a,1,VoicemailMain(${MACRO_EXTEN})


There is more to this but this should show the basics of what we use. I
store my Call Forward Numbers (CFN) in an external file. This allow me to
update the file externally (currently with a web interface but as soon as I
get the prompts recorded it will be done with an IVR) and then just reload
the extensions to activate the new numbers. Also I using SIP for pretty much
everything. Our TDM400 doesn't even have modules, it's just there for
timing. However you should be able to convert the SIP calls to ZAP calls for
you use. The internal context is included in our default context. Dialing
extension 21 calls the stdext macro. This dials the local extension first.
If not answered after 10 seconds, we check to make sure we have a phone
number to send the call out with. If not we send it on to voice mail.
Otherwise we send it to the s-CFWD. The check listed here is a very
rudimentary check but again I hope you get the idea. Next we try the call to
the CFN. If not answered in 20 seconds, then we send it to voice mail.
Finally if the user presses the star button during the attempt, we send them
on to Voicemail mail so they can check their messages.

Hopefully this helps.

John