Hi, I think ive found a bug but need someone to double check. Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100915/02bce8c9/attachment.htm
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, September 15, 2010 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bug with Realtime? Hi, I think ive found a bug but need someone to double check. Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan By reload you mean "sip reload" or just any reload in general? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100915/a09ede84/attachment.htm
On 09/15/2010 09:41 PM, Dan Journo wrote:> > Hi, > > > I think ive found a bug but need someone to double check. > > Whenever I issue a "reload" in Asterisk, any realtime extensions stop > receiving calls. > > I have to reboot the sip phones in order to get them to re-register. > > Can anyone see if they have a similar problem? > > Asterisk 1.4.32 > > Mysql realtime. > > Thanks > > Dan >Yes you loose all SIP registrations and they need to re-register to be reachable again. Don't know if this is a bug, but it's like that in 1.4 and 1.6.2. Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100915/9fdccf69/attachment-0001.htm
On 10-09-15 03:41 PM, Dan Journo wrote:> I think ive found a bug but need someone to double check. > > Whenever I issue a "reload" in Asterisk, any realtime extensions stop > receiving calls. > > I have to reboot the sip phones in order to get them to re-register. > > Can anyone see if they have a similar problem? > > Asterisk 1.4.32 > > Mysql realtime.That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. Leif.
You can do 'extensions reload' or 'ael reload' if you don't want to lose real-time sip registrations. I only reload what is needed to be reloaded instead of reloading everything. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 4:28 PM, "Leif Madsen" <leif.madsen at asteriskdocs.org> wrote: On 10-09-15 03:41 PM, Dan Journo wrote:> I think ive found a bug but need someone to double check.... That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. Leif. -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100915/927d5a0b/attachment.htm
When making an outbound call, if sip peer is not registered, first it registers itself, and then makes the call. This is why you don't see any problem dialing out. For receiving, asterisk has to wait until the sip peer registers, otherwise asterisk has nowhere to send the call. I know the pain, as I deal with the same situation. So I don't do 'reload' or 'sip reload' except if sip password (secret) has been changed, in which case I prefer to use 'sip prune realtime peer <extension>' followed by 'sip show peer <extension> load'. Most of the sip devices re-register every 60 seconds, or if they don't on a realtime network, depending upon the bandwidth, they should be made to do so. Or in some cases you can send a reboot signal to a sip device too. The bottom line is, try not to do a 'reload' as it would affect everybody else too by dropping their registrations temporarily. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-16 10:04 AM, "Peder" <peder at networkoblivion.com> wrote: A reload flushes the SIP registration database, so once you do a reload, that phones reg is gone. If the reg is set for a short period, say 60 seconds, then in 60 seconds it will re-register and work fine. Yes, it is a total pain, but this is the way it has worked since day 1 for realtime. I agree that it seems wrong and even argued that several years ago when this feature came out, but it is what it is. As someone else said, the answer is "don't do a 'reload'", do an "extensions reload" or whatever it is specific to your changes. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bo... Sent: Thursday, September 16, 2010 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussi... Subject: Re: [asterisk-users] Bug with Realtime?> That's not a bug. Only when the phone registers ...-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100916/06f5b435/attachment.htm
I am not aware of any way to do that. My question is "if you are using realtime, why are you doing a sip reload?" If you change the settings on a device in the realtime DB, just prune it and it will grab the new config the next time they re-register. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo Sent: Monday, September 20, 2010 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime?> Check the SIP debug and see what is going on. > Leif.Hi, I checked the SIP debug. As soon as I issue the RELOAD command, no SIP data gets transferred to the phone. Asterisk output: http://pastebin.com/FB675N16 Any ideas how I can do a SIP reload without losing the Sip Phones registration? Thanks Dan -- _____________________________________________________________________ -- 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