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 --
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>http://www.asterisk.org/hello
>
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> 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 --
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>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

[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*
>
> --
> _
> -- 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
>
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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*
-- 
_
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[asterisk-users] Call Forwarding Using Asterisk

[jk]

Can I do this with Asterisk,

Call comes to Asterisk Server (Master), Then master just forwards calls 
to other slave asterisk servers one by one.

Like this
Master forward 1st call to Slave 1,
Second call to Slave 2,
Third call to slave 1
Fourth call to slave 2.


Is it possible? I will appreciate if some one help me with this.

Thank you,
-Jai
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Re: [asterisk-users] Call Forwarding Using Asterisk

[jk]

Thank you Ram,
Can you give me some example, how can I  do that.

-Jk

ram wrote:

Hi
 
its possible

you need mention in the config
 
Ram


 
On 10/17/06, *jk* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Can I do this with Asterisk,

Call comes to Asterisk Server (Master), Then master just forwards
calls
to other slave asterisk servers one by one.
Like this
Master forward 1st call to Slave 1,
Second call to Slave 2,
Third call to slave 1
Fourth call to slave 2.


Is it possible? I will appreciate if some one help me with this.

Thank you,
-Jai
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