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kanth.ruban at gmail.com Guest
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Posted: Thu Sep 03, 2015 7:09 am Post subject: [asterisk-users] Call forwarding in Asterisk |
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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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Best regards,Ruban.S |
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vinicius at aittelecom... Guest
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Posted: Thu Sep 03, 2015 8:51 am Post subject: [asterisk-users] Call forwarding in Asterisk |
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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 Guest
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Posted: Fri Sep 04, 2015 2:19 am Post subject: [asterisk-users] Call forwarding in Asterisk |
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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:
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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jb_soft at trink.co.uk Guest
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Posted: Fri Sep 04, 2015 4:27 am Post subject: [asterisk-users] Call forwarding in Asterisk |
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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)
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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
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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:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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kanth.ruban at gmail.com Guest
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Posted: Fri Sep 04, 2015 4:52 am Post subject: [asterisk-users] Call forwarding in Asterisk |
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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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