[asterisk-users] 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 -- _ -- 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
Re: [asterisk-users] Restart asterisk destroy all registered SIP peers
On Fri, May 20, 2011 at 2:10 PM, satish patel 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 -- _ -- 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
Re: [asterisk-users] Restart asterisk destroy all registered SIP peers
We have polycom 501 and i am waiting since last 5 min no registration require appear. -S From: mden...@gmail.com Date: Fri, 20 May 2011 14:56:20 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers On Fri, May 20, 2011 at 2:10 PM, satish patel 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 -- _ -- 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
Re: [asterisk-users] Restart asterisk destroy all registered SIP peers
On Fri, May 20, 2011 at 3:00 PM, satish patel 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 -- _ -- 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
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. -S From: mden...@gmail.com Date: Fri, 20 May 2011 15:03:35 -0400 To: asterisk-users@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 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 -- _ -- 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
Re: [asterisk-users] Restart asterisk destroy all registered SIP peers
> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@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
Re: [asterisk-users] Restart asterisk destroy all registered SIP peers
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: ewiel...@nyigc.com > To: asterisk-users@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-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@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
Re: [asterisk-users] Restart asterisk destroy all registered SIP peers
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 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@default [7023](seb-exten) callerid="Faire Conference" <7023> accountcode="Faire Conference" mailbox=7023@default > From: ewiel...@nyigc.com > To: asterisk-users@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-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@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 -- _ -- 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
