Re: [asterisk-users] Call forwarding in Asterisk

[Julian Beach]

Hello Kantharuban,

Friday, September 4, 2015, 8:19:28 AM, you wrote:

> Thanks for your info, What is the impact of the following line in
> dialpla Dial(SIP/19201/19202,300)

It  does  not  look like a valid format. If you are trying to dial two
SIP  devices  (19201  and  19202)  with  a timeout of 300 seconds, the
command would be

Dial(SIP/19201/19202,300)  and  you might want to consider some of
the  option  Dial options depending on what you do with the call after
it has been answered.

Have  a  look  at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
for  details  of  the  dial command, and the options or have a look at
Asterisk:  The  Definitive  Guide  which  will  tell  you  more  about
Originate and Local Channels, which you might also find useful.

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html

J

-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Call forwarding in Asterisk

[Kantharuban Ruban]

Hi ,
 I have gone through the link you have sent me , there i could find the
below lines,

*Dial() together with openining Jack ports for callee*






*Nescesarry if you want to "capture" a record in leg B with SoundPatty
exten =>
_X.,n,Dial(SIP/$PROVIDER/${EXTEN},60,M(connect-jack)[macro-connect-jack]exten
=> s,1,NoOp(${CHANNEL}) ; This is leg A, skipexten =>
s,2,Set(JACK_HOOK(manipulate,i(${CHANNEL}:input),o(${CHANNEL}:output))=on)Note:
only for asterisk 1.6.x*

Could you please tell me what does it do?


On Fri, Sep 4, 2015 at 2:56 PM, Julian Beach  wrote:

> Hello Kantharuban,
>
> Friday, September 4, 2015, 8:19:28 AM, you wrote:
>
> > Thanks for your info, What is the impact of the following line in
> > dialpla Dial(SIP/19201/19202,300)
>
> It  does  not  look like a valid format. If you are trying to dial two
> SIP  devices  (19201  and  19202)  with  a timeout of 300 seconds, the
> command would be
>
> Dial(SIP/19201/19202,300)  and  you might want to consider some of
> the  option  Dial options depending on what you do with the call after
> it has been answered.
>
> Have  a  look  at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
> for  details  of  the  dial command, and the options or have a look at
> Asterisk:  The  Definitive  Guide  which  will  tell  you  more  about
> Originate and Local Channels, which you might also find useful.
>
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html
>
> J
>
> --
> Best regards,
>  Julianmailto:jb_s...@trink.co.uk
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
*Best regards,*
*Ruban.S*
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Re: [asterisk-users] Call forwarding in Asterisk

[Kantharuban Ruban]

Hi,
Thanks for your info, What is the impact of the following line in
dialplan,

Dial(SIP/19201/19202,300)





On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes 
wrote:

> 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*
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
*Best regards,*
*Ruban.S*
-- 
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