[asterisk-users] disconnect supervision

[Leandro Dardini]

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" <syscon780 at 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 at lists.**digium.com<asterisk-users-bounces at lists.digium.com>
>> [mailto:asterisk-users-**bounces at lists.digium.com<asterisk-users-bounces at 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
>>
>
> --
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