Hey Eric,

I do have qualify=yes. Am i missing something ?

[seb-exten](!)                  ; Template
type=friend
host=dynamic
context=from-sip
qualify=yes
dtmfmode=rfc2833
nat=no
cc_agent_policy=generic
cc_monitor_policy=generic

[7022](seb-exten)
callerid="Rover Conference" <7022>
accountcode="Rover Conference"
mailbox=7022@default

[7023](seb-exten)
callerid="Faire Conference" <7023>
accountcode="Faire Conference"
mailbox=7023@default



> From: [email protected]
> To: [email protected]
> Date: Fri, 20 May 2011 15:15:45 -0400
> Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP 
> peers
> 
> 
> 
> > -----Original Message-----
> > From: [email protected]
> > [mailto:[email protected]] On Behalf Of
> > satish patel
> > Sent: Friday, May 20, 2011 3:10 PM
> > To: asterisk-users
> > Subject: Re: [asterisk-users] Restart asterisk destroy all
> > registered SIP peers
> >
> > Issue is we are running customer support queue and if by
> > chance if i need to restart asterisk then they will not able
> > to get call until phone get register :(  Let me check polycom
> > default timeout and set to min.
> 
> Asterisk should cache the registrations across a restart and reboot.  I 
> belive this feature was added in 1.4.
> 
> You should not need to set a low registration timeout.  If you set it because 
> of NAT issues, setting qualify=yes will keep the translations open.
> 
> --
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