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* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150904/b226f6ed/attachment.html>
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* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150904/ed4e0dc5/attachment.html>