If you are using an analogue/sip ata, then the problem is on the ata. Run a packet capture and you'll see the invite coming from the ata without nobody using the phone...
I am typing from my mobile phone... Il giorno 11/dic/2012 18:55, "Joseph" <syscon...@gmail.com> ha scritto: > On 12/11/12 11:48, Danny Nicholas wrote: > >> In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and >> hanguponpolarityswitch lines. If they aren't present, the default values >> are being used. If they are, tweak them and restart asterisk and dahdi. >> I >> do this - service asterisk stop; service dahdi restart; service asterisk >> start. >> > > I'm not using dahdi.conf I'm using extension.conf sip.conf with analog > AudioCodes gateway FXO/FXS > > -- > Joseph > > >> -----Original Message----- >> From: >> asterisk-users-bounces@lists.**digium.com<asterisk-users-boun...@lists.digium.com> >> [mailto:asterisk-users-**boun...@lists.digium.com<asterisk-users-boun...@lists.digium.com>] >> On Behalf Of Joseph >> Sent: Tuesday, December 11, 2012 11:44 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] disconnect supervision >> >> On 12/11/12 11:30, Danny Nicholas wrote: >> >>> Could be, but I'd check the "easier to fix" polarity settings. >>> >> >> How do I do that? >> >> Notice, that this channel hang-up/disconnect does not happen all the time, >> only once a while could be once a day or once a week. >> >> -- >> Joseph >> > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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