asterisk users - May 2011 - Restart asterisk destroy all registered SIP peers

Hi Guys!

This is strange issue with 1.8 I have restarted my asterisk and it destroy all
registered SIP peers now only solution is i manually reboot all phones to get
them register back. I have never seen issue like this before. Any idea what
would be the issue ?

Thanks
S
 		 	   		  
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On Fri, May 20, 2011 at 2:10 PM, satish patel <satish_lx at hotmail.com>
wrote:
>  Hi Guys!
>
> This is strange issue with 1.8 I have restarted my asterisk and it destroy
> all registered SIP peers now only solution is i manually reboot all phones
> to get them register back. I have never seen issue like this before. Any
> idea what would be the issue ?
>
> Thanks
> S
>
Shouldn't the phones re-register on their own?  Mine do it every few
minutes.

-M
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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.

-S

From: mdeneen at gmail.com
Date: Fri, 20 May 2011 15:03:35 -0400
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP	peers



On Fri, May 20, 2011 at 3:00 PM, satish patel <satish_lx at hotmail.com>
wrote:







We have polycom 501 and i am waiting since last 5 min no registration require
appear.

-S


With Polycom 321 you can poke around the menus -- one of them has a countdown
timer which will show you when the next registration happens.


-M 

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> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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.

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 at default

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


> From: EWieling at nyigc.com
> To: asterisk-users at lists.digium.com
> Date: Fri, 20 May 2011 15:15:45 -0400
> Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP
peers
> 
> 
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] 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.
> 
> --
> _____________________________________________________________________
> -- 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
 		 	   		  
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There is a fix https://issues.asterisk.org/view.php?id=19318

--
Sent from my iPhone

On May 20, 2011, at 4:40 PM, satish patel <satish_lx at hotmail.com>
wrote:
> 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 at default
>
> [7023](seb-exten)
> callerid="Faire Conference" <7023>
> accountcode="Faire Conference"
> mailbox=7023 at default
>
>
>
> > From: EWieling at nyigc.com
> > To: asterisk-users at lists.digium.com
> > Date: Fri, 20 May 2011 15:15:45 -0400
> > Subject: Re: [asterisk-users] Restart asterisk destroy all  
> registered SIP peers
> >
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] 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.
> >
> > --
> >  
> _____________________________________________________________________
> > -- 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
> --
> _____________________________________________________________________
> -- 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
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