Re: [asterisk-users] Call forwarding in Asterisk

[Vinicius Fontes]

You might want to use the Originate() application instead. Check its usage
by issuing the command 'core show application originate' on Asterisk CLI.

2015-09-03 9:09 GMT-03:00 Kantharuban Ruban :

> Hello Group,
>
> I have a requirement to dialout some external number, once the
> call is answered the same has to be forwarded to an Internal Queue.
>
> Please help me.
>
> I have tried calling with two SIP end point forwarding , even that is not
> working,
>
> My dial plan line is , Dial(SIP/19201/19202,300)
>
>
> --
> *Best regards,*
> *Ruban.S*
>
> --
> _
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[asterisk-users] Can ChanSpy offer private dialog?

[Michael Munger]

I have a customer who needs to use chanspy to :

1.   Interrupt a call and whisper to the local extension

2.   Allow the person on the local extension to reply to the interrupter, 
but not have the remote party hear it.

Narrative:

John is on the phone with a customer. Doris receives an urgent call from John's 
mother. Doris uses to break into the conversation John is having with a 
customer. She tells John: "Your mother is on the phone. It's urgent." John 
replies to Doris: "OK, I'll call her in two minutes." The customer on the 
remote end heard none of this.

The ChanSpy whisper mode appears to do half of this: Doris can whisper to John, 
but John cannot acknowledge her and let her know he heard her and what his 
course / plan of action will be.

Does ChanSpy support this or is there another application I should be using?

Michael Munger, dCAP, MCPS, MCNPS, MBSS
High Powered Help, Inc.
Microsoft Certified Professional
Microsoft Certified Small Business Specialist
Digium Certified Asterisk Professional
mich...@highpoweredhelp.com

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[asterisk-users] Call forwarding in Asterisk

[Kantharuban Ruban]

Hello Group,

I have a requirement to dialout some external number, once the
call is answered the same has to be forwarded to an Internal Queue.

Please help me.

I have tried calling with two SIP end point forwarding , even that is not
working,

My dial plan line is , Dial(SIP/19201/19202,300)


-- 
*Best regards,*
*Ruban.S*
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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