In article <cajujwtig7yzk4+kb3c6sdu6zhb_+vwsg-oy0pibw0maeeed...@mail.gmail.com>, SamyGo <[email protected]> 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 <[email protected]>wrote: > > > In article < > > caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com>, > > Mehdi Rahimi <[email protected]> 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: [email protected] - http://www.softins.co.uk > > Play: [email protected] - 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: [email protected] - http://www.softins.co.uk Play: [email protected] - 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
