I need your help in implementing the following scenario: A certain extension will ring two sip phones simultaneously and when one of them answers, the other keeps ringing until it answers too, and then all three (the caller and the other two) are immediately placed in a conference room (same room for all three). Can we do it?
Hello, look at Page application Regards ----- Original Message ----- From: "Thanasis" <thanasis at asyr.hopto.org> To: <asterisk-users at lists.digium.com> Sent: Monday, October 31, 2011 4:59 PM Subject: [asterisk-users] custom automated meeting>I need your help in implementing the following scenario: > > A certain extension will ring two sip phones simultaneously and when one > of them answers, the other keeps ringing until it answers too, and then > all three (the caller and the other two) are immediately placed in a > conference room (same room for all three). > > Can we do it? > > -- > _____________________________________________________________________ > -- 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
I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable?
One way to do this (there are probably more and better ways). Incoming call
to 123456789 launches meetme(1234,b(connecta.agi))
Connecta.agi calls lines B and C and connects them to meetme(1234).
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Thanasis
Sent: Tuesday, November 01, 2011 1:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] custom automated meeting
I just want to make two specific sip phone sets to ring together, when
someone dials a specific incoming extension. And then, when each of the
ringed sets answers, to be placed immediately into meeting session with the
caller together with the other phone set.
Here is exactly what I mean:
Person A dials 123456789. Asterisk routes the incoming call and rings sip
phones B and C. Person B answers phone B and starts talking with person A,
while phone C keeps ringing. A minute later, and while A and B are still
talking together, person C answers phone C, and starts talking with A and B
together (that is aromatically all being placed in the same conference
session).
Is that doable?
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on 11/01/2011 03:25 PM Danny Nicholas wrote the following:> One way to do this (there are probably more and better ways). Incoming call > to 123456789 launches meetme(1234,b(connecta.agi)) > Connecta.agi calls lines B and C and connects them to meetme(1234).Thanks, but could you be more elaborate please? Where can I find connecta.agi ?> > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Thanasis > Sent: Tuesday, November 01, 2011 1:58 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] custom automated meeting > > I just want to make two specific sip phone sets to ring together, when > someone dials a specific incoming extension. And then, when each of the > ringed sets answers, to be placed immediately into meeting session with the > caller together with the other phone set. > > Here is exactly what I mean: > > Person A dials 123456789. Asterisk routes the incoming call and rings sip > phones B and C. Person B answers phone B and starts talking with person A, > while phone C keeps ringing. A minute later, and while A and B are still > talking together, person C answers phone C, and starts talking with A and B > together (that is aromatically all being placed in the same conference > session). > > Is that doable?
There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin directory by developing it yourself. On Tue, Nov 1, 2011 at 6:57 PM, Thanasis <thanasis at asyr.hopto.org> wrote:> on 11/01/2011 03:25 PM Danny Nicholas wrote the following: > > One way to do this (there are probably more and better ways). Incoming > call > > to 123456789 launches meetme(1234,b(connecta.agi)) > > Connecta.agi calls lines B and C and connects them to meetme(1234). > > Thanks, but could you be more elaborate please? > Where can I find connecta.agi ? > > > > > -----Original Message----- > > From: asterisk-users-bounces at lists.digium.com > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Thanasis > > Sent: Tuesday, November 01, 2011 1:58 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] custom automated meeting > > > > I just want to make two specific sip phone sets to ring together, when > > someone dials a specific incoming extension. And then, when each of the > > ringed sets answers, to be placed immediately into meeting session with > the > > caller together with the other phone set. > > > > Here is exactly what I mean: > > > > Person A dials 123456789. Asterisk routes the incoming call and rings sip > > phones B and C. Person B answers phone B and starts talking with person > A, > > while phone C keeps ringing. A minute later, and while A and B are still > > talking together, person C answers phone C, and starts talking with A > and B > > together (that is aromatically all being placed in the same conference > > session). > > > > Is that doable? > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111101/109203f1/attachment-0001.htm>
Although if you dig through the archives you can find a good cross-section of AGI samples. Check the Asterisk Cookbook wikis as well. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sammy Govind Sent: Tuesday, November 01, 2011 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin directory by developing it yourself. On Tue, Nov 1, 2011 at 6:57 PM, Thanasis <thanasis at asyr.hopto.org> wrote: on 11/01/2011 03:25 PM Danny Nicholas wrote the following:> One way to do this (there are probably more and better ways). Incomingcall> to 123456789 launches meetme(1234,b(connecta.agi)) > Connecta.agi calls lines B and C and connects them to meetme(1234).Thanks, but could you be more elaborate please? Where can I find connecta.agi ?> > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Thanasis > Sent: Tuesday, November 01, 2011 1:58 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] custom automated meeting > > I just want to make two specific sip phone sets to ring together, when > someone dials a specific incoming extension. And then, when each of the > ringed sets answers, to be placed immediately into meeting session withthe> caller together with the other phone set. > > Here is exactly what I mean: > > Person A dials 123456789. Asterisk routes the incoming call and rings sip > phones B and C. Person B answers phone B and starts talking with person A, > while phone C keeps ringing. A minute later, and while A and B are still > talking together, person C answers phone C, and starts talking with A andB> together (that is aromatically all being placed in the same conference > session). > > Is that doable?-- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111101/edcd6233/attachment.htm>
You need simple dialplan of four steps:
same =>n,Set(conf_name=conf-${RAND(1,1000)})
same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =>n,MeetMe(${conf_name},dFI1xAC)
same =>n,Noop(do post conference stuff)
2011/10/31 Thanasis <thanasis at asyr.hopto.org>:> I need your help in implementing the following scenario:
>
> A certain extension will ring two sip phones simultaneously and when one
> of them answers, the other keeps ringing until it answers too, and then
> all three (the caller and the other two) are immediately placed in a
> conference room (same room for all three).
>
> Can we do it?
>
> --
> _____________________________________________________________________
> -- 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
>
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:> You need simple dialplan of four steps: > same =>n,Set(conf_name=conf-${RAND(1,1000)}) > same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x) > same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x) > same =>n,MeetMe(${conf_name},dFI1xAC) > same =>n,Noop(do post conference stuff) >Thanks! What is the meaning of the options dFI1xAC passed to app,MeetMe,${conf_name} ? Where can I find them described please?> > 2011/10/31 Thanasis <thanasis at asyr.hopto.org>: >> I need your help in implementing the following scenario: >> >> A certain extension will ring two sip phones simultaneously and when one >> of them answers, the other keeps ringing until it answers too, and then >> all three (the caller and the other two) are immediately placed in a >> conference room (same room for all three). >> >> Can we do it? >>
Type in asterisk CLI>core show application meetme or google "asterisk cmd meetme" simple? On Tue, Nov 1, 2011 at 10:33 PM, Thanasis <thanasis at asyr.hopto.org> wrote:> on 11/01/2011 05:41 PM Yaroslav Panych wrote the following: > > You need simple dialplan of four steps: > > same =>n,Set(conf_name=conf-${RAND(1,1000)}) > > same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x) > > same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x) > > same =>n,MeetMe(${conf_name},dFI1xAC) > > same =>n,Noop(do post conference stuff) > > > > Thanks! > What is the meaning of the options dFI1xAC passed to > app,MeetMe,${conf_name} ? > Where can I find them described please? > > > > > 2011/10/31 Thanasis <thanasis at asyr.hopto.org>: > >> I need your help in implementing the following scenario: > >> > >> A certain extension will ring two sip phones simultaneously and when one > >> of them answers, the other keeps ringing until it answers too, and then > >> all three (the caller and the other two) are immediately placed in a > >> conference room (same room for all three). > >> > >> Can we do it? > >> > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111102/3a085a06/attachment.htm>
on 11/02/2011 07:44 AM Sammy Govind wrote the following:> core show application meetmeThanks! (I am new to asterisk, and just learning, so forgive my dumb questions)
on 10/31/2011 11:59 PM Thanasis wrote the following:> I need your help in implementing the following scenario: > > A certain extension will ring two sip phones simultaneously and when one > of them answers, the other keeps ringing until it answers too, and then > all three (the caller and the other two) are immediately placed in a > conference room (same room for all three). > > Can we do it?FWIW, using call files: Here is the relevant section of the dialplan: exten => 300,1,Noop(creating conference) same => n,Set(conf_name=conf-${RAND(1,1000)}) same => n,System(/etc/asterisk/scripts/callgenerator SIP/dev1 ${conf_name}) same => n,System(/etc/asterisk/scripts/callgenerator SIP/dev2 ${conf_name}) same => n,MeetMe(${conf_name},dFI1xAC) same => n,Noop(do post conference stuff) ... and here is the script /etc/asterisk/scripts/callgenerator: #!/bin/bash PHONE=$(echo $1 |cut -f2 -d"/") ROOM=$2 echo "Channel: $1" > /var/spool/asterisk/tmp/${PHONE}.call echo "Application: MeetMe" >> /var/spool/asterisk/tmp/${PHONE}.call echo "Data: ${ROOM},dFI1x" >> /var/spool/asterisk/tmp/${PHONE}.call mv -f /var/spool/asterisk/tmp/${PHONE}.call /var/spool/asterisk/outgoing PS: Thanks much to Yaroslav for his help :)