Deka, Rajib IN MAA SL
2011-Apr-20 09:50 UTC
[asterisk-users] No voice in MeetMe for SIP with
Thanks a lot Tony and Dhaval for your much appreciable suggestions.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com
Date: Wed, 20 Apr 2011 13:55:25 +0530
From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>
Subject: Re: [asterisk-users] No voice in MeetMe for SIP with
AGI_BACKGROUND
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTikgRHjCVJhBC097S8n9YM66VWp=QA at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
hey try with app_rpt in asterisk
regards
dhaval
On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield <tony at
softins.co.uk>wrote:
> In article <
> 2658E54B540D284981EA57E6A549EA70ABD1FDF921 at
INBLRK77M1MSX.in002.siemens.net
> >,
> Deka, Rajib IN MAA SL <rajib.deka at siemens.com> wrote:
> >
> > The requirement is little complicated as it is H/W specific.
> > Basically we are integrating a radio gateway (SIP) with asterisk. The
> gateway will be
> > connected to a meetme room, so that any operator (with IP phone
> registered as SIP user to
> > asterisk) can login to the room and listen to radio communications and
> talk.
> >
> > Using a PTT button someone can talk on a radio channel. Once someone
> presses the PTT button
> > a SIP MESSAGE is sent to the gateway with a string as payload to
enable
> half duplex
> > communication. So, we were planning to run an AGI script with meetme
> (AGI_BACKGROUND) to
> > receive the MESSAGE (using AGI command 'RECEIVE TEXT') from
both ends and
> to generate a
> > VarSet AMI event.
> >
> > Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)->
SIP:MESSAGE
> -> radio gateway
> > And vise versa.
> >
> > Any suggestions on the above scenario.
>
> I don't think it can be done without making modifications to Asterisk.
>
> The first thing I would do, if you haven't done so already, would be to
> try it without MeetMe:
>
> Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE
->
> radio gateway
>
> If that works, then it would suggest that the SIP MESSAGE is
> successfully getting translated into an ast_frame, which is then getting
> translated back into a SIP MESSAGE. If that is not happening, you might
> need to add some code to chan_sip.c to do those steps.
>
> Once Asterisk is converting the message to and from an ast_frame, the
> next step would be to add some code to app_meetme.c in the conf_run()
> function, to pass those frames through, in the same way as DTMF frames
> get passed through when the F option is enabled.
>
> Presumably the messages represent PTT PRESS and PTT RELEASE. You will
> need to decide what to do if you have two operators connected and they
> both press the PTT.
>
> You might also need to automatically unmute or mute the operator
> channel when their PTT is pressed or released. That could also be done
> within the MeetMe code.
>
> There may be other approaches too...
>
> Hope this helps!
> Tony
> --
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
>
> --
> _____________________________________________________________________
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Thank You.
is your problem solved or not On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL < rajib.deka at siemens.com> wrote:> Thanks a lot Tony and Dhaval for your much appreciable suggestions. > > Regards, > Rajib > > Rajib Deka > SIEMENS Ltd. > Robert V Chandran Tower, First Floor, West Wing, > #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. > www.siemens.com > > Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com > > Date: Wed, 20 Apr 2011 13:55:25 +0530 > From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com> > Subject: Re: [asterisk-users] No voice in MeetMe for SIP with > AGI_BACKGROUND > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <BANLkTikgRHjCVJhBC097S8n9YM66VWp=QA at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > hey try with app_rpt in asterisk > > regards > dhaval > > On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield <tony at softins.co.uk > >wrote: > > > In article < > > > 2658E54B540D284981EA57E6A549EA70ABD1FDF921 at INBLRK77M1MSX.in002.siemens.net > > >, > > Deka, Rajib IN MAA SL <rajib.deka at siemens.com> wrote: > > > > > > The requirement is little complicated as it is H/W specific. > > > Basically we are integrating a radio gateway (SIP) with asterisk. The > > gateway will be > > > connected to a meetme room, so that any operator (with IP phone > > registered as SIP user to > > > asterisk) can login to the room and listen to radio communications and > > talk. > > > > > > Using a PTT button someone can talk on a radio channel. Once someone > > presses the PTT button > > > a SIP MESSAGE is sent to the gateway with a string as payload to enable > > half duplex > > > communication. So, we were planning to run an AGI script with meetme > > (AGI_BACKGROUND) to > > > receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends > and > > to generate a > > > VarSet AMI event. > > > > > > Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> > SIP:MESSAGE > > -> radio gateway > > > And vise versa. > > > > > > Any suggestions on the above scenario. > > > > I don't think it can be done without making modifications to Asterisk. > > > > The first thing I would do, if you haven't done so already, would be to > > try it without MeetMe: > > > > Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE -> > > radio gateway > > > > If that works, then it would suggest that the SIP MESSAGE is > > successfully getting translated into an ast_frame, which is then getting > > translated back into a SIP MESSAGE. If that is not happening, you might > > need to add some code to chan_sip.c to do those steps. > > > > Once Asterisk is converting the message to and from an ast_frame, the > > next step would be to add some code to app_meetme.c in the conf_run() > > function, to pass those frames through, in the same way as DTMF frames > > get passed through when the F option is enabled. > > > > Presumably the messages represent PTT PRESS and PTT RELEASE. You will > > need to decide what to do if you have two operators connected and they > > both press the PTT. > > > > You might also need to automatically unmute or mute the operator > > channel when their PTT is pressed or released. That could also be done > > within the MeetMe code. > > > > There may be other approaches too... > > > > Hope this helps! > > Tony > > -- > > Tony Mountifield > > Work: tony at softins.co.uk - http://www.softins.co.uk > > Play: tony at mountifield.org - http://tony.mountifield.org > > > > -- > > _____________________________________________________________________ > > -- 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 > > > > > Important notice: This e-mail and any attachment there to contains > corporate proprietary information. If you have received it by mistake, > please notify us immediately by reply e-mail and delete this e-mail and its > attachments from your system. > Thank You. > > -- > _____________________________________________________________________ > -- 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/20110420/5c81b667/attachment.htm>
In article <BANLkTik8+63ghnM6WTT2M7kH1PBXgSkYdg at mail.gmail.com>, DHAVAL INDRODIYA <dhaval.it01034 at gmail.com> wrote:> > is your problem solved or notIt will take a lot more time than that to try out the suggestions!> On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL < > rajib.deka at siemens.com> wrote: > > > Thanks a lot Tony and Dhaval for your much appreciable suggestions. > > > > Regards, > > Rajib > > > > Rajib Deka > > SIEMENS Ltd. > > Robert V Chandran Tower, First Floor, West Wing, > > #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. > > www.siemens.com-- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org