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 <vinicius at
aittelecom.com.br>
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 <kanth.ruban at
gmail.com>:
>
>> 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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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&SIP/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, Julian mailto:jb_soft at trink.co.uk
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
<http://github.com/Motiejus/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 <jb_soft at trink.co.uk>
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&SIP/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,
> Julian mailto:jb_soft at 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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