Hi Tony, Thank you for your attention , and appreciate your contribution . You are right we can not do anything till the caller hangup BUT how can we prevent to hearing DTMF when someone else is trying on another extension ? to clearance : someone calls (from landlines os mobile , no difference) and our AGI has executed and after some processes finish and hangup , but the caller has not hungup yet and till then if i pickup my extension and try to call , that caller who has not hungup the call yet can hear DTMF and that's a problem and some conflict.
Regards, Mehdi On Tue, Sep 18, 2012 at 5:35 PM, Tony Mountifield <t...@softins.co.uk> wrote: > In article > <cajujwtig7yzk4+kb3c6sdu6zhb_+vwsg-oy0pibw0maeeed...@mail.gmail.com>, > SamyGo <govoi...@gmail.com> wrote: >> >> So basically the FXO cards configurations need to be tweaked i.e >> hanguponpolarityinverse=yes etc. >> Since this is a Hangup request initiated by the SIP client, Asterisk then >> atleast it should close all the media streams and channel should get >> deleted. >> Keeping an eye on BYE : *CLI> "sip set debug on" Then make this call and >> see if a SIP BYE method is triggered properly and appears on screen. >> More likely you need to look into you dahdi configs. >> >> Thanks, >> Sammy > > I think you are misunderstanding the OP's issue. > > Hangup on polarity reversal would only apply if Asterisk were making the > call to a phone and wanted to me informed if the phone (called party) > hung up. > > The OP's situation is different. The extension below is invoked by an > INCOMING call to Asterisk, and he is then trying to hang up that call > from the Asterisk (called) end. > > If the caller is a SIP phone, that is fine, as either end can hang up. > > Hi problem is that when the incoming call is via his FXO port, the PSTN > does not drop the call when the Asterisk end hangs up the FXO line. In > this scenario there is on SIP involved. The problem is that the PSTN > will not drop the call when the called party on an analogue line hangs > up, until after a long timeout. There is usually no solution to this. > > Cheers > Tony > >> On Tue, Sep 18, 2012 at 2:03 PM, Tony Mountifield <t...@softins.co.uk>wrote: >> >> > In article < >> > caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com>, >> > Mehdi Rahimi <mrm.ci...@gmail.com> wrote: >> > > Hi all, >> > > >> > > I need to handle a problem from AGI please guide me >> > > >> > > in extensions_custom.conf : >> > > >> > > exten => s,1,Answer >> > > exten => s,n,AGI(hang.php) >> > > exten => s,n,Hangup >> > > >> > > in hang.php : >> > > >> > > #!/usr/bin/php -q >> > > <? >> > > set_time_limit(30); >> > > require('phpagi.php'); >> > > error_reporting(E_ALL); >> > > $agi = new AGI(); >> > > $agi->answer(); >> > > $agi->say_number('10000'); >> > > $agi->hangup(); >> > > ?> >> > > >> > > >> > > calling from an extension has no problem but whenever i use landline >> > > or mobile it can not hangup the call and the caller has to hangup the >> > > call. >> > >> > In the UK phone network, and I suspect in many other countries too, for >> > analogue lines it is the caller who holds the call open. For example in >> > a call between two normal analogue phones, the called party can hangup >> > their phone, and then within a short while pick it up again (or another >> > phone on the same line) and the caller is still there. Hanging up the >> > called phone does not clear down the call until after quite a long >> > timeout (a couple of minutes perhaps). >> > >> > In your above example with Asterisk connected to an analogue line with an >> > FXO card, Asterisk is the called party, and is therefore unable to clear >> > down the line forcibly. This is not an Asterisk or AGI problem but a PSTN >> > one. >> > >> > Cheers >> > Tony >> > -- >> > Tony Mountifield >> > Work: t...@softins.co.uk - http://www.softins.co.uk >> > Play: t...@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 >> > >> >> -=-=-=-=-=- >> [Alternative: text/html] >> -=-=-=-=-=- >> -=-=-=-=-=- >> >> -- >> _____________________________________________________________________ >> -- 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 >> -=-=-=-=-=- > > > -- > Tony Mountifield > Work: t...@softins.co.uk - http://www.softins.co.uk > Play: t...@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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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