Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051211/3dd12f8d/attachment.htm
It doesn't "execute" a command upon registration. What it does is insert a simple noop into the extension and context at priority 1 so the extension then becomes active. For example: Before I register: Exten => 145,2,Dial(SIP/jcolp_cisco1) When I register it then turns into: Exten => 145,1,Noop() Exten => 145,2,Dial(SIP/jcolp_cisco1) This means when I'm registered, a person can call me but when I'm not - the extension is useless and calling it does nothing and doesn't attempt to call me. Joshua Colp _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang Sent: Sunday, December 11, 2005 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Regexten Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051211/5ac73ef7/attachment.htm
Ah ok. Thanks. I was hoping to use it to 'replicate' registrations from one Asterisk system to another. Darn it. -----Original Message----- From: Joshua Colp [mailto:joshnet@nbnet.nb.ca] Sent: Sunday, December 11, 2005 1:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Regexten It doesn't "execute" a command upon registration. What it does is insert a simple noop into the extension and context at priority 1 so the extension then becomes active. For example: Before I register: Exten => 145,2,Dial(SIP/jcolp_cisco1) When I register it then turns into: Exten => 145,1,Noop() Exten => 145,2,Dial(SIP/jcolp_cisco1) This means when I'm registered, a person can call me but when I'm not - the extension is useless and calling it does nothing and doesn't attempt to call me. Joshua Colp _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang Sent: Sunday, December 11, 2005 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Regexten Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051211/2a7234b8/attachment.htm
On 12/11/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> The misconception seems to be that it allows you to execute a command upon > registration from a SIP UA. Even the O'Reilly TFOT book erroneously states > that this is what it is for. After reading the developer discussion though, > it definitely seems to be broken. Is it fixed yet?And I quote from page 227, "Asterisk will dynamically create and destroy a NoOp at priority 1 for the extension. All actions to be performed upon registration should start at priority 2." Leif Madsen. http://www.oreilly.com/catalog/asterisk http://www.leifmadsen.com
No. It doesn't work that way. -----Original Message----- From: Leif Madsen [mailto:asterisk.leif.madsen@gmail.com] Sent: Sun 12/11/2005 4:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Regexten On 12/11/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > The misconception seems to be that it allows you to execute a command upon > registration from a SIP UA. Even the O'Reilly TFOT book erroneously states > that this is what it is for. After reading the developer discussion though, > it definitely seems to be broken. Is it fixed yet? And I quote from page 227, "Asterisk will dynamically create and destroy a NoOp at priority 1 for the extension. All actions to be performed upon registration should start at priority 2." Leif Madsen. http://www.oreilly.com/catalog/asterisk http://www.leifmadsen.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4542 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051211/019334b4/attachment.bin
So, After about 6 months I think I've finally worked out how regexten works. And... it doesn't look like it will serve my purpose. I'm hope I'm wrong, otherwise what a letdown! It also means that DUNDI becomes unusable as a HA mechanism. Let me see if I understand its function correctly tho... My sip.conf: [2944093] ... regcontext=sip_autoreg regexten=2944093 extensions.conf: [sip_autoreg] exten => 2944093,2,Answer exten => 2944093,3,Dial(SIP/2944093) exten => 2944093,4,Hangup Phone comes up. Asterisk sees this: *CLI> -- Saved useragent "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 However, I don't have a NoOp() priority 1. [ Context 'sip_autoreg' created by 'pbx_config' ] '2944093' => 2. Answer() [pbx_config] 3. Dial(SIP/2944093) [pbx_config] 4. Hangup() [pbx_config] Why?
Thanks for the reply, but still no luck. sip.conf: [2944093] ... regcontext=sip_autoreg regexten=2944093 extensions.conf: [From_OneEighty] include => sip_autoreg [sip_autoreg] exten => 2944093,2,Answer exten => 2944093,3,Dial(SIP/2944093) exten => 2944093,4,Hangup Phone comes up, and registers: *CLI> -- Saved useragent "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 Dialplan before and after shows the same thing: '2944093' => 2. Answer() [pbx_config] 3. Dial(SIP/2944093) [pbx_config] 4. Hangup() [pbx_config] Doug.> -----Original Message----- > From: Luigi Rizzo [mailto:rizzo@icir.org] > Sent: Thursday, March 16, 2006 4:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] regexten > > > On Thu, Mar 16, 2006 at 03:55:18PM -0700, Douglas Garstang wrote: > > So, After about 6 months I think I've finally worked out > how regexten works. > > > > And... it doesn't look like it will serve my purpose. I'm > hope I'm wrong, otherwise what a letdown! It also means that > DUNDI becomes unusable as a HA mechanism. > > as unfortunate as it can be, at the moment you need to use > a different context for entries generated by 'regexten' > and entries statically configured. Then you have to include > one into the other so you can match all entries on a call. > > I opened a bug on mantis some time ago, indicating two possible > fixes. Hopefully this will be resolved sooner or later. > > cheers > luigi > > > Let me see if I understand its function correctly tho... > > > > My sip.conf: > > [2944093] > > ... > > regcontext=sip_autoreg > > regexten=2944093 > > > > extensions.conf: > > [sip_autoreg] > > exten => 2944093,2,Answer > > exten => 2944093,3,Dial(SIP/2944093) > > exten => 2944093,4,Hangup > > > > Phone comes up. Asterisk sees this: > > *CLI> -- Saved useragent > "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > > > However, I don't have a NoOp() priority 1. > > > > [ Context 'sip_autoreg' created by 'pbx_config' ] > > '2944093' => 2. Answer() > [pbx_config] > > 3. Dial(SIP/2944093) > [pbx_config] > > 4. Hangup() > [pbx_config] > > > > Why? > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> -----Original Message----- > From: Luigi Rizzo [mailto:rizzo@icir.org] > Sent: Thursday, March 16, 2006 4:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] regexten > > > On Thu, Mar 16, 2006 at 04:27:10PM -0700, Douglas Garstang wrote: > > Thanks for the reply, but still no luck. > > > > sip.conf: > > [2944093] > > ... > > regcontext=sip_autoreg > > regexten=2944093 > > > > extensions.conf: > > [From_OneEighty] > > include => sip_autoreg > > > > [sip_autoreg] > > exten => 2944093,2,Answer > > exten => 2944093,3,Dial(SIP/2944093) > > exten => 2944093,4,Hangup > > I repeat: > > > > as unfortunate as it can be, at the moment you need to use > > > a different context for entries generated by 'regexten' > > you still have static entries in context [sip_autoreg] > in extensions.conf, and use the same context for > entries generated by regexten - regcontext=sip_autoregSorry? Say again? If you don't mind, you could you have another go at explaining that? That really didn't make sense to me. I do have a different context for entries generated by regexten... it's [sip_autoreg] Doug> cheers > luigi > > > Phone comes up, and registers: > > *CLI> -- Saved useragent > "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > > > Dialplan before and after shows the same thing: > > '2944093' => 2. Answer() > [pbx_config] > > 3. Dial(SIP/2944093) > [pbx_config] > > 4. Hangup() > [pbx_config] > > > > Doug. > > > > > > > -----Original Message----- > > > From: Luigi Rizzo [mailto:rizzo@icir.org] > > > Sent: Thursday, March 16, 2006 4:14 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] regexten > > > > > > > > > On Thu, Mar 16, 2006 at 03:55:18PM -0700, Douglas Garstang wrote: > > > > So, After about 6 months I think I've finally worked out > > > how regexten works. > > > > > > > > And... it doesn't look like it will serve my purpose. I'm > > > hope I'm wrong, otherwise what a letdown! It also means that > > > DUNDI becomes unusable as a HA mechanism. > > > > > > as unfortunate as it can be, at the moment you need to use > > > a different context for entries generated by 'regexten' > > > and entries statically configured. Then you have to include > > > one into the other so you can match all entries on a call. > > > > > > I opened a bug on mantis some time ago, indicating two possible > > > fixes. Hopefully this will be resolved sooner or later. > > > > > > cheers > > > luigi > > > > > > > Let me see if I understand its function correctly tho... > > > > > > > > My sip.conf: > > > > [2944093] > > > > ... > > > > regcontext=sip_autoreg > > > > regexten=2944093 > > > > > > > > extensions.conf: > > > > [sip_autoreg] > > > > exten => 2944093,2,Answer > > > > exten => 2944093,3,Dial(SIP/2944093) > > > > exten => 2944093,4,Hangup > > > > > > > > Phone comes up. Asterisk sees this: > > > > *CLI> -- Saved useragent > > > "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > > > > > > > However, I don't have a NoOp() priority 1. > > > > > > > > [ Context 'sip_autoreg' created by 'pbx_config' ] > > > > '2944093' => 2. Answer() > > > [pbx_config] > > > > 3. Dial(SIP/2944093) > > > [pbx_config] > > > > 4. Hangup() > > > [pbx_config] > > > > > > > > Why? > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Well, I finally got it to work. Such a shame I can't use it. I didn't realise it until I'd expended all the effort, but this approach doesn't give you a HA asterisk solution. If the server that the phone is registered to goes down, no Asterisk system knows the location of the phones that where registered to it. If you had 3 Asterisk boxes, 1/3 of your users suddenly can't receive calls. Not HA! -----Original Message----- From: Luigi Rizzo [mailto:rizzo@icir.org] Sent: Thu 3/16/2006 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] regexten On Thu, Mar 16, 2006 at 04:27:10PM -0700, Douglas Garstang wrote: > Thanks for the reply, but still no luck. > > sip.conf: > [2944093] > ... > regcontext=sip_autoreg > regexten=2944093 > > extensions.conf: > [From_OneEighty] > include => sip_autoreg > > [sip_autoreg] > exten => 2944093,2,Answer > exten => 2944093,3,Dial(SIP/2944093) > exten => 2944093,4,Hangup I repeat: > > as unfortunate as it can be, at the moment you need to use > > a different context for entries generated by 'regexten' you still have static entries in context [sip_autoreg] in extensions.conf, and use the same context for entries generated by regexten - regcontext=sip_autoreg cheers luigi > Phone comes up, and registers: > *CLI> -- Saved useragent "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > Dialplan before and after shows the same thing: > '2944093' => 2. Answer() [pbx_config] > 3. Dial(SIP/2944093) [pbx_config] > 4. Hangup() [pbx_config] > > Doug. > > > > -----Original Message----- > > From: Luigi Rizzo [mailto:rizzo@icir.org] > > Sent: Thursday, March 16, 2006 4:14 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] regexten > > > > > > On Thu, Mar 16, 2006 at 03:55:18PM -0700, Douglas Garstang wrote: > > > So, After about 6 months I think I've finally worked out > > how regexten works. > > > > > > And... it doesn't look like it will serve my purpose. I'm > > hope I'm wrong, otherwise what a letdown! It also means that > > DUNDI becomes unusable as a HA mechanism. > > > > as unfortunate as it can be, at the moment you need to use > > a different context for entries generated by 'regexten' > > and entries statically configured. Then you have to include > > one into the other so you can match all entries on a call. > > > > I opened a bug on mantis some time ago, indicating two possible > > fixes. Hopefully this will be resolved sooner or later. > > > > cheers > > luigi > > > > > Let me see if I understand its function correctly tho... > > > > > > My sip.conf: > > > [2944093] > > > ... > > > regcontext=sip_autoreg > > > regexten=2944093 > > > > > > extensions.conf: > > > [sip_autoreg] > > > exten => 2944093,2,Answer > > > exten => 2944093,3,Dial(SIP/2944093) > > > exten => 2944093,4,Hangup > > > > > > Phone comes up. Asterisk sees this: > > > *CLI> -- Saved useragent > > "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > > > > > However, I don't have a NoOp() priority 1. > > > > > > [ Context 'sip_autoreg' created by 'pbx_config' ] > > > '2944093' => 2. Answer() > > [pbx_config] > > > 3. Dial(SIP/2944093) > > [pbx_config] > > > 4. Hangup() > > [pbx_config] > > > > > > Why? > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 10222 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/c240cee8/attachment.bin
Luigi, I got to it work in the configuration I posted. Funny bug... Anyway I had to put regcontext in the globals section of sip.conf, not against a specific user. Thanks for your help. Unfortunately, since this approach with DUNDi only has phones registered to a single Asterisk box, it isn't much of a HA solution. Having phones unreachable for the re-registration period isn't an option. Doug -----Original Message----- From: Luigi Rizzo [mailto:rizzo@icir.org] Sent: Thu 3/16/2006 11:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] regexten On Thu, Mar 16, 2006 at 04:41:42PM -0700, Douglas Garstang wrote: > > > > -----Original Message----- > > From: Luigi Rizzo [mailto:rizzo@icir.org] > > Sent: Thursday, March 16, 2006 4:34 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] regexten > > > > > > On Thu, Mar 16, 2006 at 04:27:10PM -0700, Douglas Garstang wrote: > > > Thanks for the reply, but still no luck. > > > > > > sip.conf: > > > [2944093] > > > ... > > > regcontext=sip_autoreg > > > regexten=2944093 > > > > > > extensions.conf: > > > [From_OneEighty] > > > include => sip_autoreg > > > > > > [sip_autoreg] > > > exten => 2944093,2,Answer > > > exten => 2944093,3,Dial(SIP/2944093) > > > exten => 2944093,4,Hangup > > > > I repeat: > > > > > > as unfortunate as it can be, at the moment you need to use > > > > a different context for entries generated by 'regexten' > > > > you still have static entries in context [sip_autoreg] > > in extensions.conf, and use the same context for > > entries generated by regexten - regcontext=sip_autoreg > Sorry? Say again? If you don't mind, you could you have another go at explaining that? That really didn't make sense to me. > I do have a different context for entries generated by regexten... it's [sip_autoreg] no you have only one context - see above your email - both your static context and your regexten context are named 'sip_autoreg' and asterisk cannot handle this case. As i said, it is a bug (at least the fact that it doesn't send an error message) and i reported it already. There is a workaround, which is using contexts with different names and including the 'dynamic' one into the static one, so the problem is not terribly critical. Hope now it is clear enough. bye luigi > Doug > > > cheers > > luigi > > > > > Phone comes up, and registers: > > > *CLI> -- Saved useragent > > "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > > > > > Dialplan before and after shows the same thing: > > > '2944093' => 2. Answer() > > [pbx_config] > > > 3. Dial(SIP/2944093) > > [pbx_config] > > > 4. Hangup() > > [pbx_config] > > > > > > Doug. > > > > > > > > > > -----Original Message----- > > > > From: Luigi Rizzo [mailto:rizzo@icir.org] > > > > Sent: Thursday, March 16, 2006 4:14 PM > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: Re: [Asterisk-Users] regexten > > > > > > > > > > > > On Thu, Mar 16, 2006 at 03:55:18PM -0700, Douglas Garstang wrote: > > > > > So, After about 6 months I think I've finally worked out > > > > how regexten works. > > > > > > > > > > And... it doesn't look like it will serve my purpose. I'm > > > > hope I'm wrong, otherwise what a letdown! It also means that > > > > DUNDI becomes unusable as a HA mechanism. > > > > > > > > as unfortunate as it can be, at the moment you need to use > > > > a different context for entries generated by 'regexten' > > > > and entries statically configured. Then you have to include > > > > one into the other so you can match all entries on a call. > > > > > > > > I opened a bug on mantis some time ago, indicating two possible > > > > fixes. Hopefully this will be resolved sooner or later. > > > > > > > > cheers > > > > luigi > > > > > > > > > Let me see if I understand its function correctly tho... > > > > > > > > > > My sip.conf: > > > > > [2944093] > > > > > ... > > > > > regcontext=sip_autoreg > > > > > regexten=2944093 > > > > > > > > > > extensions.conf: > > > > > [sip_autoreg] > > > > > exten => 2944093,2,Answer > > > > > exten => 2944093,3,Dial(SIP/2944093) > > > > > exten => 2944093,4,Hangup > > > > > > > > > > Phone comes up. Asterisk sees this: > > > > > *CLI> -- Saved useragent > > > > "PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 2944093 > > > > > > > > > > However, I don't have a NoOp() priority 1. > > > > > > > > > > [ Context 'sip_autoreg' created by 'pbx_config' ] > > > > > '2944093' => 2. Answer() > > > > [pbx_config] > > > > > 3. Dial(SIP/2944093) > > > > [pbx_config] > > > > > 4. Hangup() > > > > [pbx_config] > > > > > > > > > > Why? > > > > > _______________________________________________ > > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > > > Asterisk-Users mailing list > > > > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 13058 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060317/32393158/attachment.bin
A few days ago I didn't realise that the phones only registered to one Asterisk box. If I did, I wouldn't have spent hours today trying to get something working. Is this really the best pure Asterisk HA solution that's out there right now? I think it needs a little work. Doug. -----Original Message----- From: Peter Bowyer [mailto:peeebeee@gmail.com] Sent: Thu 3/16/2006 11:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] regexten On 17/03/06, Douglas Garstang <dgarstang@oneeighty.com> wrote: > Well, I finally got it to work. Such a shame I can't use it. I didn't realise it until I'd expended all the effort, but this approach doesn't give you a HA asterisk solution. If the server that the phone is registered to goes down, no Asterisk system knows the location of the phones that where registered to it. If you had 3 Asterisk boxes, 1/3 of your users suddenly can't receive calls. Not HA! It does if you combine this with an IAX switch or DUNDI, and phones which re-register fast. Did you read how this was explained to you a few days ago? -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org VoIP: *5048707000@sipbroker.com FWD: **275*5048707000 VoipTalk: **473*5048707000 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4766 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060317/0aab8588/attachment.bin
Any thing better than five-nines is a much better sale pitch. LoL. BTW. My Cisco 7960s alows for a entry of a backup proxy. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tony Mountifield Sent: Friday, March 17, 2006 4:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: regexten In article <645FEC31A18FE54A8721500CDD55A7B602C1538F@mail.oneeighty.com>, Douglas Garstang <dgarstang@oneeighty.com> wrote:> A few days ago I didn't realise that the phones only registered to one> Asterisk box. If I did, I wouldn't have spent hours today trying to > get something working.I would have thought that it was a function of the phone how many servers it could register with, not a limitation of Asterisk. There are phones that can have more than one "presence" (line). I have never played with one, but I wonder if it is possible to get one to register as the same extensions number on two different servers.> Is this really the best pure Asterisk HA solution that's out there > right now? I think it needs a little work.I've been watching your trials and tribulations, and I wonder how often you are expecting an Asterisk box to go down? I have some Asterisk boxes and other Linux boxes with 200-300 days uptime. If you have your phones register on a 2-min interval, then provided Asterisk keeps running for (2min / 0.001%) = 139 days or more, then you have "five-nines" (99.999%) availability. And don't forget that if the phones re-register at 2-min intervals randomly spread, the AVERAGE downtime after a failure would only be one minute. It then becomes a question of whether it is cost-effective to strive for even better than that. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
> -----Original Message----- > From: Gabriel Afana [mailto:asterisk@gafana.com] > Sent: Friday, March 17, 2006 1:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: regexten > > > He does have a good point! > > > ----- Original Message ----- > From: "Tony Mountifield" <tony@softins.clara.co.uk> > To: <asterisk-users@lists.digium.com> > Sent: Friday, March 17, 2006 1:04 AM > Subject: [Asterisk-Users] Re: regexten > > > > In article > <645FEC31A18FE54A8721500CDD55A7B602C1538F@mail.oneeighty.com>, > > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > A few days ago I didn't realise that the phones only > registered to one > > > Asterisk box. If I did, I wouldn't have spent hours today > trying to > > > get something working. > > > > I would have thought that it was a function of the phone how many > > servers it could register with, not a limitation of Asterisk.It's a function of both. When the phone sucessfully registers with an Asterisk system, it stops. The phone therefore only registers with a single Asterisk system, and the same problem still remains. The phone is only registered to one Asterisk system. If that Asterisk system goes down, none of the phones registered to that system can receive calls. However, if by some magic the phone _could_ send another registration, Asterisk would accept it. I've yet to see a phone however, that will send another registration if the first suceeds. Why would it?> > > > There are phones that can have more than one "presence" > (line). I have > > never played with one, but I wonder if it is possible to get one to > > register as the same extensions number on two different servers. > > > > > Is this really the best pure Asterisk HA solution that's out there > > > right now? I think it needs a little work. > > > > I've been watching your trials and tribulations, and I > wonder how often > > you are expecting an Asterisk box to go down? I have some > Asterisk boxes > > and other Linux boxes with 200-300 days uptime.It doesn't matter how often I expect them to go down. You have to plan for the worst. People are used to say, being unable to access a web page for a few minutes. Heck, they might not even notice. However, when someone's phone system goes down, you can bet your bottom dollar that they will notice pretty damn fast, and they expect it to always be working.> > > > If you have your phones register on a 2-min interval, then provided > > Asterisk keeps running for (2min / 0.001%) = 139 days or more, then > > you have "five-nines" (99.999%) availability. > > > > And don't forget that if the phones re-register at 2-min intervals > > randomly spread, the AVERAGE downtime after a failure would > only be one > > minute. > > > > It then becomes a question of whether it is cost-effective > to strive for > > even better than that. > > > > Cheers > > Tony > > -- > > Tony Mountifield > > Work: tony@softins.co.uk - http://www.softins.co.uk > > Play: tony@mountifield.org - http://tony.mountifield.org > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
The polycom doesn't need to re-register... it's already registered on all three Asterisk systems. :)> -----Original Message----- > From: Gabriel Afana [mailto:asterisk@gafana.com] > Sent: Friday, March 17, 2006 3:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: regexten > > > > The Polycom phones can be configured with secondary and tertiary SIP > > servers in the event of server failure. Thus, the Polycoms will > > automatically register with another server if their primary > SIP server > > goes offline. > > Where can you set a tertiary server? Its only primary and secondary - > unless you are talking about using DNS-SRV, in which case you > can add as > many as you like. > > However, this is not entirely true. According to Polycom, > the phone will > *not* send another registration request to the secondary > phone in the event > you pick up the phone and make a call and the primary server > has failed. It > will only make the call but not register with it. The phone will only > re-register when the registartion expires. According to Polycom and > Asterisk/Digium, set the expiration to 1 minute and for th > emost part you > should be "ok". > > - Gabe > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Yes.> -----Original Message----- > From: Gabriel Afana [mailto:asterisk@gafana.com] > Sent: Friday, March 17, 2006 3:22 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: regexten > > > Your talking about with SER right? > > > ----- Original Message ----- > From: "Douglas Garstang" <dgarstang@oneeighty.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Friday, March 17, 2006 2:13 PM > Subject: RE: [Asterisk-Users] Re: regexten > > > The polycom doesn't need to re-register... it's already > registered on all > three Asterisk systems. :) > > > -----Original Message----- > > From: Gabriel Afana [mailto:asterisk@gafana.com] > > Sent: Friday, March 17, 2006 3:04 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Re: regexten > > > > > > > The Polycom phones can be configured with secondary and > tertiary SIP > > > servers in the event of server failure. Thus, the Polycoms will > > > automatically register with another server if their primary > > SIP server > > > goes offline. > > > > Where can you set a tertiary server? Its only primary and > secondary - > > unless you are talking about using DNS-SRV, in which case you > > can add as > > many as you like. > > > > However, this is not entirely true. According to Polycom, > > the phone will > > *not* send another registration request to the secondary > > phone in the event > > you pick up the phone and make a call and the primary server > > has failed. It > > will only make the call but not register with it. The > phone will only > > re-register when the registartion expires. According to Polycom and > > Asterisk/Digium, set the expiration to 1 minute and for th > > emost part you > > should be "ok". > > > > - Gabe > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >