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Kind of like DISA but not I think...

Posted by IVSCOMM on Tue, 09/09/2008

OK Here is the description. I am John Q Salesguy and I am driving to cleveland to sell a hotdog vending cart to a one eyed retired beachcomber. (has nothing to do with my request but it's kind of fun) He has his office phone forwarded to his cell phone. And just as he is leaving Toledo he gets a call from his supplier that the hot dog cart is done but some information is needed before it can be shipped. The information needed is not in the car but in the office with his sales assistant. I need that person to be able to enter a code (like a flash) and get internal dialtone and transfer the caller to the assistant. Thereby making the cellphone a true remote extension. The dial tone should only be offered from forwarded extension. not from outbound calling.

Anyone, Bueller...Bueller...

shawn


Submitted by IVSCOMM on Tue, 09/09/2008 Permalink

This assumes that the forwarded call is somehow monitored by the Asterisk/Thirdlane.

Submitted by mattdarnell on Mon, 09/15/2008 Permalink

That should definitely work, all calls, unless otherwise programmed, are routed through asterisk.

Check this out in features.conf:

; Note that the DTMF features listed below only work when two channels have answered and are bridged together.

; They can not be used while the remote party is ringing or in progress. If you require this feature you can use

; chan_local in combination with Answer to accomplish it.

[featuremap]

;blindxfer => #1 ; Blind transfer (default is #)

;disconnect => *0 ; Disconnect (default is *)

;automon => *1 ; One Touch Record a.k.a. Touch Monitor

;atxfer => *2 ; Attended transfer

;parkcall => #72 ; Park call (one step parking)

;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor

I have done it before where a call came to my cellular phone, I entered the transfer digits, got internal dialtone, dialed a three digit extension to transfer.

-Matt

Submitted by dozment on Mon, 09/15/2008 Permalink

I'm not getting DTMF back to my Asterisk server when I try this. I'm logging DTMF, so I can see when it happens. I use G.711, no reinvites, and RFC2833 everywhere.

Submitted by eeman on Sat, 09/20/2008 Permalink

in order to use dtmf for transfers the t and T options have to be given to the Dial application

t - Allow the called party to transfer the calling party by sending the

DTMF sequence defined in features.conf.

T - Allow the calling party to transfer the called party by sending the

DTMF sequence defined in features.conf.

are you forwarding from the user portal in pbxmanager or are you forwarding from your sip phone?