[asterisk-users] Calls drop after 20 seconds

[Peder]

Like the poster below said, do a sip debug on a call and see which end sends
the bye message or ends the call and go from there.  That should give you
some sort of clue as to who is having a timer issue.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Wednesday, April 21, 2010 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls drop after 20 seconds

Alejandro Recarey schrieb:
> Doug, thanks for the help, already looked it up, but it does not seem
> to be a NAT issue (which is what most posters suggest when googling)
>
> Danny, those are billsec durations, the call has been established and
> media is being passed for 20 seconds.
>
> Thanks again!
>
> Alex
>
>   
Hi,

How do you dial the users? direct with the peername or something like 
exten at ipofpeer ?

i know this problem when dialing a patton ISDN ata without an extension. 
The call is established but when the T1 sip timeout fires the call gets 
disconnected. Maybe you could do some sip debugging and watch for resend 
sip messages.

best regards

steve

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Stefan Schmidt
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