[asterisk-users] Transfer Call to Cell Phone

[OCOSA ListAcct]

works perfect,,,,,,,,,,,,Thanks

--
Otis 



John Faubion wrote:
>> 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
>
>
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