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[asterisk-users] Call forwarding in Asterisk


 
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kanth.ruban at gmail.com
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PostPosted: Thu Sep 03, 2015 7:09 am    Post subject: [asterisk-users] Call forwarding in Asterisk Reply with quote

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) 




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vinicius at aittelecom...
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PostPosted: Thu Sep 03, 2015 8:51 am    Post subject: [asterisk-users] Call forwarding in Asterisk Reply with quote

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@gmail.com (kanth.ruban@gmail.com)>:
Quote:
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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kanth.ruban at gmail.com
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PostPosted: Fri Sep 04, 2015 2:19 am    Post subject: [asterisk-users] Call forwarding in Asterisk Reply with quote

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@aittelecom.com.br (vinicius@aittelecom.com.br)> wrote:
Quote:
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@gmail.com (kanth.ruban@gmail.com)>:


Quote:
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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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jb_soft at trink.co.uk
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PostPosted: Fri Sep 04, 2015 4:27 am    Post subject: [asterisk-users] Call forwarding in Asterisk Reply with quote

Hello Kantharuban,

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

Quote:
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@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:
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kanth.ruban at gmail.com
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PostPosted: Fri Sep 04, 2015 4:52 am    Post subject: [asterisk-users] Call forwarding in Asterisk Reply with quote

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 calleeNescesarry if you want to "capture" a record in leg B with SoundPattyexten => _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@trink.co.uk (jb_soft@trink.co.uk)> wrote:
Quote:
Hello Kantharuban,

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

Quote:
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@trink.co.uk (jb_soft@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






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Best regards,Ruban.S
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