[asterisk-users] Calls drop after 20 seconds

[Alejandro Recarey]

> Vieri
> Check out the "early dial" feature in the Grandstream products (if you enabled it)
> and play with the "pedantic" option.

thanks, already made sure I use pedantic=no and earlydial is off in my GW

> 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.

That is my next step, its just so hard to reproduce while debugging!


> Stefan
> 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.

I don't understand, all of my calls are inbound and terminated with
different voip carriers, so I am not sure how that will work. I always
dial dst at ipofcarrier. Will debug!

> Ishfaq
> Upgrade phones to latest/most stable firmware
> Upgrade routers to latest/most stable firmware

This has definetly helped with other problems in the past, so I
reccomend it to anybody

Thank you so much for all of your help / time guys!