[asterisk-users] AGI HANGUP PROBLEM
Mehdi Rahimi
mrm.cisco at gmail.com
Tue Sep 18 09:13:39 CDT 2012
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 <tony at softins.co.uk> wrote:
> In article <CAJUJwtig7YZk4+kB3C6SdU6zhB_+vWSg-OY0PiBw0MaEeed4Hw at mail.gmail.com>,
> SamyGo <govoiper at 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 <tony at softins.co.uk>wrote:
>>
>> > In article <
>> > CAEhsOWEanTzTYOebdoBjchOeSZhfk_z9SigAUJSiJ15XX-uEtA at mail.gmail.com>,
>> > Mehdi Rahimi <mrm.cisco at 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: 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
>> >
>>
>> -=-=-=-=-=-
>> [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: 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
More information about the asterisk-users
mailing list