Hi, 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 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 >
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