So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned.
On 20/12/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned.Off you go to another product then. Close the door on the way out. -- 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
Digium needs people like me, if they read this list that is. They sure don't seem to be able to make real-world functionality decisions on their own. -----Original Message----- From: Peter Bowyer [mailto:peeebeee@gmail.com] Sent: Tuesday, December 20, 2005 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions On 20/12/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned.Off you go to another product then. Close the door on the way out. -- 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
I don't think the bounties are worth the costs asociated with contracts, legal fees, international boundaries etc. -----Original Message----- From: Darren Wiebe [mailto:darren@aleph-com.net] Sent: Tuesday, December 20, 2005 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions http://www.voip-info.org/wiki/view/Asterisk+bounty Darren Wiebe darren@aleph-com.net Douglas Garstang wrote:>Digium needs people like me, if they read this list that is. They sure don't seem to be able to make real-world functionality decisions on their own. > >-----Original Message----- >From: Peter Bowyer [mailto:peeebeee@gmail.com] >Sent: Tuesday, December 20, 2005 3:18 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] SIP Subscriptions > > >On 20/12/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > >>So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. >> >> > >Off you go to another product then. Close the door on the way out. > > >-- >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 >_______________________________________________ >--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 > >-- Darren Wiebe darren@aleph-com.net Aleph Communications ASTPP - Open Source Voip Billing & Calling Cards www.aleph-com.net/astpp _______________________________________________ --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
Doesn't realtime allow you to manipulate users without ever having to issue a reload command? I'd expect the command 'reload' to flush any existing information from memory and use with freshly loaded information anyway, and that goes for any application, not just asterisk. On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> > So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose > all your SIP subscriptions. Nice. Basically that means the use of hints and > subscriptions in a production environment is a completely impossible. > Awesome. Considering traditional phone users have come to expect this > functionality, it leaves a lot to be desired as far as Asterisk is > concerned. > _______________________________________________ > --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 -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/41604a9c/attachment.htm
Realtime, as stated by Digium, does not work with sip users. This isn't related to sip subscriptions. -----Original Message----- From: Gary Reuter [mailto:gary.reuter@gmail.com] Sent: Tuesday, December 20, 2005 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions Doesn't realtime allow you to manipulate users without ever having to issue a reload command? I'd expect the command 'reload' to flush any existing information from memory and use with freshly loaded information anyway, and that goes for any application, not just asterisk. On 12/20/05, Douglas Garstang < dgarstang@oneeighty.com> wrote: So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. _______________________________________________ --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 -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/4d188209/attachment.htm
You know, there isn't even a 'sip delete subscription' command available. If there was, it might be possible to write some third party scripts which interact via the Manager Interface to control subscriptions. Without even a delete function though, it's pretty much impossible to do anything at all with this. (Asterisk will just keep piling up the subscriptions, presumabley until it runs out of memory and crashes). -----Original Message----- From: Douglas Garstang Sent: Tuesday, December 20, 2005 3:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: SIP Subscriptions So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned.
I don't think it's an expiry issue. When a phone reboots before the expiry, it sends a new subscription to Asterisk. Asterisk doesn't remove the old one. If subscriptions where keyed like registrations, the new one would just overwrite the old one. -----Original Message----- From: BJ Weschke [mailto:bweschke@gmail.com] Sent: Tuesday, December 20, 2005 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: SIP Subscriptions On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> You know, there isn't even a 'sip delete subscription' command available. If there was, it might be possible to write some third party scripts which interact via the Manager Interface to control subscriptions. Without even a delete function though, it's pretty much impossible to do anything at all with this. (Asterisk will just keep piling up the subscriptions, presumabley until it runs out of memory and crashes). >I believe the patch to correct the issue with "old" subscriptions not expiring when new one comes in has already been applied to the /trunk and /branches/1.2 codebase in svn. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Dang. So we still have a subscription buildup. I didn't even realise Asterisk was ignoring the expiry. Pardon me while I cry! -----Original Message----- From: BJ Weschke [mailto:bweschke@gmail.com] Sent: Tuesday, December 20, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: SIP Subscriptions On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> I don't think it's an expiry issue. When a phone reboots before the expiry, it sends a new subscription to Asterisk. Asterisk doesn't remove the old one. If subscriptions where keyed like registrations, the new one would just overwrite the old one. >No. You're right. It's not an expiry issue, but a patch was applied within the past week that does "expire" the old one now when the new one comes in. :) -- Bird's The Word Technologies, Inc. http://www.btwtech.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
No... I'm using standard 1.2.1. I could... and I may.... I'll keep this in mind. But... for now I'm not sure if I want to introduce more variables. -----Original Message----- From: BJ Weschke [mailto:bweschke@gmail.com] Sent: Tuesday, December 20, 2005 4:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: SIP Subscriptions On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> Dang. So we still have a subscription buildup. I didn't even realise Asterisk was ignoring the expiry. Pardon me while I cry! >Are you running a version of the code later than this past Friday? http://svn.digium.com/view/asterisk?rev=7513&view=rev Give it a shot. It may turn your frown upside down. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Google tells me that the first message from you on this list is on Dec 6, while there might be an error in that, I doubt that Google is off by more than a week. (http://www.google.com/search?hl=en&q=%22Douglas+Garstang%22+site%3Adigium.com&btnG=Google+Search) We have done well overhere on this list without your bashing and ranting, you can go back to where you are coming from. Sleep with your Panasonic/Avaya/Nortel/Toshiba/NEC or whatever other crapy (err good) system you were using until Dec/6, and leave us alone. I'm sure between one of those mentioned you can find one that will not time out after 30 seconds when you set it to 60 using queues, or one that still uses CVS, or one that will load ztdummy without reading docs on how to do it, or one that will not loose it's subscriptions while doing a reload, or one that will play the announce file you specify and not 0, or one that supports ACD from Polycom phones, or one that does support Realtime to share sip registration, or one that comes with real good docs on using realtime, or one that updates the fullcontact field when using sip in conjuction with SER (but that shouldn't be too relevant), or one that when doing 'select * from sip_buddies where username=<called-username>'. doesnt' fail periodically.... even Under certain circumstances that you don't yet understand, or one that does use the values of the DB and not AstDB, or one that doesn't have HA problems when your so lazy, or one that has such a big user support group as this. Douglas Garstang, please stop ranting bashing swearing and cursing when thing don't go your way, it wont get you anywhere (minus my ignore list). If you have a problem: * Some end users of asterisk (users and installers, that know it for a bit longer than since Dec/6, and have done bigger installations for PRODUCTION environments than 120 users), will gladly help you with a config problem but NOT if you act the way you did since Dec/6 * Developers will gladly fix any bugs you find, and even listen to suggestion for improvements, but not with this attitude. * New features, you can air them out on the list (respecting others), and if enough interest exists a bounty can be set up. I don't think that *ANYBODY* on this list owes *YOU* anything, and as the other poster said, close the door on the way out, I will just add one thing: trip on the way out so you can't ever come back. Now about your stupid post. On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned."/completely impossible"/ Yeah I agree, because if subscriptions and hints done work, your calls don't go thru. So I think you should dump it. "/traditional phone users/" Wow, do you mean traditiona phone users of Verizon? or of a key system? as Verizon doesn't realy offer that service, nor does any other PBX (although some will allow you limited functions in this area, unless you are paying tons of money for a receptionists console, it will never come to what you can get done with Asterisk and FOP). "/it leaves a lot to be desired as far as Asterisk is concerned./" You remind me of the guy that didn't want to take out that spoon of his coffee cup. Leave and that way rid yourself of that curse called Asterisk, and us of that curse called Douglas Garstang.
> C F <shmaltz@gmail.com> wrote: > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > Digium needs people like me, if they read this list that is. They sure > don't seem to be able to make real-world functionality decisions on > their own. > > > > You are so right, how did we do without you the last few > years???????????By living on baked beans and cabbage. PaulH
Robert La Ferla
2005-Dec-20 20:10 UTC
[Asterisk-Users] low audio volume on recorded .wav voicemail messages
The audio volume of voicemail messages (msgNNN.wav) is rather low. Is there a parameter/option to adjust gain? In my voicemail.conf, I use these formats: format=wav49|gsm|wav Maybe I should use a different format?
But the beans and cabbage remind me more of duggie. PaulH> Aaron Daniel <amdtech@shsu.edu> wrote: > > OH, and don't forget pizza! > > Aaron > On Dec 20, 2005, at 9:02 PM, pdhales@optusnet.com.au wrote: > > >> C F <shmaltz@gmail.com> wrote: > >> > >> On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > >>> Digium needs people like me, if they read this list that is. They > >>> sure > >> don't seem to be able to make real-world functionality decisions on > >> their own. > >>> > >> > >> You are so right, how did we do without you the last few > >> years??????????? > > > > By living on baked beans and cabbage. > > > > PaulH > > _______________________________________________ > > --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
Good grief. Your perceptions of reality are quite warped. It seems you have a problem with me asking a lot of valid questions. Why exactly is that? -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 12/20/2005 7:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions Google tells me that the first message from you on this list is on Dec 6, while there might be an error in that, I doubt that Google is off by more than a week. (http://www.google.com/search?hl=en&q=%22Douglas+Garstang%22+site%3Adigium.com&btnG=Google+Search) We have done well overhere on this list without your bashing and ranting, you can go back to where you are coming from. Sleep with your Panasonic/Avaya/Nortel/Toshiba/NEC or whatever other crapy (err good) system you were using until Dec/6, and leave us alone. I'm sure between one of those mentioned you can find one that will not time out after 30 seconds when you set it to 60 using queues, or one that still uses CVS, or one that will load ztdummy without reading docs on how to do it, or one that will not loose it's subscriptions while doing a reload, or one that will play the announce file you specify and not 0, or one that supports ACD from Polycom phones, or one that does support Realtime to share sip registration, or one that comes with real good docs on using realtime, or one that updates the fullcontact field when using sip in conjuction with SER (but that shouldn't be too relevant), or one that when doing 'select * from sip_buddies where username=<called-username>'. doesnt' fail periodically.... even Under certain circumstances that you don't yet understand, or one that does use the values of the DB and not AstDB, or one that doesn't have HA problems when your so lazy, or one that has such a big user support group as this. Douglas Garstang, please stop ranting bashing swearing and cursing when thing don't go your way, it wont get you anywhere (minus my ignore list). If you have a problem: * Some end users of asterisk (users and installers, that know it for a bit longer than since Dec/6, and have done bigger installations for PRODUCTION environments than 120 users), will gladly help you with a config problem but NOT if you act the way you did since Dec/6 * Developers will gladly fix any bugs you find, and even listen to suggestion for improvements, but not with this attitude. * New features, you can air them out on the list (respecting others), and if enough interest exists a bounty can be set up. I don't think that *ANYBODY* on this list owes *YOU* anything, and as the other poster said, close the door on the way out, I will just add one thing: trip on the way out so you can't ever come back. Now about your stupid post. On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. "/completely impossible"/ Yeah I agree, because if subscriptions and hints done work, your calls don't go thru. So I think you should dump it. "/traditional phone users/" Wow, do you mean traditiona phone users of Verizon? or of a key system? as Verizon doesn't realy offer that service, nor does any other PBX (although some will allow you limited functions in this area, unless you are paying tons of money for a receptionists console, it will never come to what you can get done with Asterisk and FOP). "/it leaves a lot to be desired as far as Asterisk is concerned./" You remind me of the guy that didn't want to take out that spoon of his coffee cup. Leave and that way rid yourself of that curse called Asterisk, and us of that curse called Douglas Garstang. _______________________________________________ --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: 7082 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/043715cb/attachment.bin
Seems someone has some anger management issues. As I just stated in a previous post, it seems you have issues with me asking valid questions. I'm not sure why that is. The long email you rattled off with all my questions where quite valid. Your issue with that is.....? -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 12/20/2005 7:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > I don't think the bounties are worth the costs asociated with contracts, legal fees, international boundaries etc. Right, Avaya is. This guy is a Fuc*ing retard. Douglas, did you see a doc the last few days??????????? _______________________________________________ --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: 4254 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/501d9a1a/attachment.bin
I don't know what the rules are for this list, but it wouldn't be much of a stretch to assume that personal attacks are grounds for removal. While we're at it, why does my reluctance to deal with contracts, legal fees and international boundaries make me a 'fuc*ing retard'? -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 12/20/2005 7:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > I don't think the bounties are worth the costs asociated with contracts, legal fees, international boundaries etc. Right, Avaya is. This guy is a Fuc*ing retard. Douglas, did you see a doc the last few days??????????? _______________________________________________ --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
Since when where you in a position to come to the conclusion that my complaints are idle? My complains and questions are valid, just like anyone else's. -----Original Message----- From: Aaron Daniel [mailto:amdtech@shsu.edu] Sent: Tue 12/20/2005 9:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions This email is a perfect example. "Your issue with that is...?" Your sarcastic attitude is grating on a lot of people's nerves, which is what his email was about. If you can't put in constructive criticism with a better attitude and less "asterisk sucks", then go somewhere else with your complaints. If you don't like the product, fix it, or don't use it, but quit annoying everyone else with your idle complaints because your system isn't working. Aaron On Dec 20, 2005, at 9:52 PM, Douglas Garstang wrote: > Seems someone has some anger management issues. As I just stated in > a previous post, it seems you have issues with me asking valid > questions. I'm not sure why that is. The long email you rattled off > with all my questions where quite valid. Your issue with that is.....? > > -----Original Message----- > From: C F [mailto:shmaltz@gmail.com] > Sent: Tue 12/20/2005 7:33 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: [Asterisk-Users] SIP Subscriptions > > > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > I don't think the bounties are worth the costs asociated with > contracts, legal fees, international boundaries etc. > > Right, Avaya is. > > This guy is a Fuc*ing retard. > Douglas, did you see a doc the last few days??????????? > _______________________________________________ > --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 > > > <winmail.dat> > _______________________________________________ > --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: 6690 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/8e1d7b42/attachment.bin
Well, I know I will be attacked for saying this, as cursed are those who say anything bad about Asterisk, but for an application that is 'enterprise-grade' as Digium tourts, it has several flaws that IMHO disqualify it as 'enterprise-grade'. I mean, really, c'mon... think about it... an application that doesn't support the use of a database for common location/registration information is hardly 'enterprise-grade'. Just off the top of my head, other disqualifiers would be: 1. SIP subscriptions are stored in memory and cleared when you do a 'reload'. So, if you make any configuration changes and 'reload' you lose all your BLF lights. People take this stuff for granted and expect it to work. 2. No common SIP registration information. Not even using realtime with SIP users, which doesn't work, there's no way outside this to share location info between more than one (ie 'enterprise-grade') Asterisk systems. 3. The 'Dial' application seem to have very limited ability to be able to determine what SIP response it gets back from a peer. "Not Found", "Busy", "Moved" etc. I know Asterisk isn't a SIP proxy, but without the ability to check the SIP message status in a dial, it makes redundancy very very difficult. Redundancy is normally an important part of 'enterprise-grade'. Without this, how do you get upstream redundancy? I have something working right now, but it isn't pretty! 4. DNS SRV lookups aren't implemented properly. Another important part of redundancy and 'enterprise-grade' software. Before you all go ripping my head off and defecating down my neck for saying anything bad about Asterisk (well CF can, I expect that), this is just my opinion based on real world frustrations with Asterisk. These are intended to be constructive criticisms. Maybe some from Digium will read this email and it will make a 0.001% contribution towards some of these things being fixed. Oh, and no... I can't switch to another solution. The decision was made above my head to go with Asterisk. It's my job to make it do all that 'enterprise-grade' stuff. Doug. -----Original Message----- From: pdhales@optusnet.com.au [mailto:pdhales@optusnet.com.au] Sent: Tue 12/20/2005 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: Re: [Asterisk-Users] SIP Subscriptions > C F <shmaltz@gmail.com> wrote: > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > Digium needs people like me, if they read this list that is. They sure > don't seem to be able to make real-world functionality decisions on > their own. > > > > You are so right, how did we do without you the last few > years??????????? By living on baked beans and cabbage. PaulH _______________________________________________ --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
I hope you are just having a breakdown and aren't normally like this.> > Well, I know I will be attacked for saying this, as cursed are thosewho> say anything bad about Asterisk, but for an application that is > 'enterprise-grade' as Digium tourts, it has several flaws that IMHO > disqualify it as 'enterprise-grade'. I mean, really, c'mon... thinkabout> it... an application that doesn't support the use of a database forcommon> location/registration information is hardly 'enterprise-grade'. Justoff> the top of my head, other disqualifiers would be: > > 1. SIP subscriptions are stored in memory and cleared when you do a > 'reload'. So, if you make any configuration changes and 'reload' youlose> all your BLF lights. People take this stuff for granted and expect itto> work. > > 2. No common SIP registration information. Not even using realtimewith> SIP users, which doesn't work, there's no way outside this to share > location info between more than one (ie 'enterprise-grade') Asterisk > systems. > > 3. The 'Dial' application seem to have very limited ability to be ableto> determine what SIP response it gets back from a peer. "Not Found","Busy",> "Moved" etc. I know Asterisk isn't a SIP proxy, but without theability to> check the SIP message status in a dial, it makes redundancy very very > difficult. Redundancy is normally an important part of'enterprise-grade'.> Without this, how do you get upstream redundancy? I have somethingworking> right now, but it isn't pretty! > > 4. DNS SRV lookups aren't implemented properly. Another important partof> redundancy and 'enterprise-grade' software. > > Before you all go ripping my head off and defecating down my neck for > saying anything bad about Asterisk (well CF can, I expect that), thisis> just my opinion based on real world frustrations with Asterisk. Theseare> intended to be constructive criticisms. > > Maybe some from Digium will read this email and it will make a 0.001% > contribution towards some of these things being fixed. Oh, and no... I > can't switch to another solution. The decision was made above my headto> go with Asterisk. It's my job to make it do all that'enterprise-grade'> stuff. > > Doug. > > > -----Original Message----- > From: pdhales@optusnet.com.au [mailto:pdhales@optusnet.com.au] > Sent: Tue 12/20/2005 8:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: Re: [Asterisk-Users] SIP Subscriptions > > > > > C F <shmaltz@gmail.com> wrote: > > > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > Digium needs people like me, if they read this list that is. > They sure > > don't seem to be able to make real-world functionalitydecisions> on > > their own. > > > > > > > You are so right, how did we do without you the last few > > years??????????? > > By living on baked beans and cabbage. > > PaulH > _______________________________________________ > --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 >
I respectfully disagree that my complaints are idle. Digium as 'Digium? is the creator and primary developer of Asterisk?, ' state that it is 'enterprise-grade '. I feel that the things I have complained about disqualify it as being enterprise-grade. That's not idle. If it's enterprise-grade, these things should be supported and should work. I realise that Digium is saying the business edition is enterprise grade, but if you look at it from a functionality perspective, business edition at about version 1.0.9, which has even _less_ features than 1.2.1. Oh, and no I haven't looked at the realtime code. Firstly because digium have told me it would take a single person the better part of a year to fix, and secondly I'm not a software developer (not in C anyway). Not everyone who uses open source code is a software engineer. Yes, I did suggest it to Digium directly.... which is when they mentioned it would take a year or so to fix. -----Original Message----- From: Aaron Daniel [mailto:amdtech@shsu.edu] Sent: Tue 12/20/2005 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions Your complaints are idle, because you provide no real solution to the problem, other than "digium needs better qa" and various other comments about how crappy asterisk is. I've been working with asterisk for 8-9 months now, have been utterly pissed off at the system for weeks, but it takes patience and work. And this list has provided much support in the time I've spent working on our system, because most people on here have provided a situation they're in, and the solutions that follow are usually pretty good (and some are pretty neat). People aren't going to want to help if you continuously post about your distaste for the product without any real solution. Seriously though, have you looked through the realtime code to see if there's anything you can do to make it better? Have you looked at the subscription functions to see if there's anything in there you can work on to add a remove? Or even suggest it be worked on to people with possibly more time or skills on the project? Without those, I'd definitely say your complaints are idle. On Dec 20, 2005, at 10:21 PM, Douglas Garstang wrote: > Since when where you in a position to come to the conclusion that > my complaints are idle? My complains and questions are valid, just > like anyone else's. > > -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Tue 12/20/2005 9:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: [Asterisk-Users] SIP Subscriptions > > > > This email is a perfect example. > > "Your issue with that is...?" > > Your sarcastic attitude is grating on a lot of people's nerves, which > is what his email was about. If you can't put in constructive > criticism with a better attitude and less "asterisk sucks", then go > somewhere else with your complaints. If you don't like the product, > fix it, or don't use it, but quit annoying everyone else with your > idle complaints because your system isn't working. > > Aaron > > On Dec 20, 2005, at 9:52 PM, Douglas Garstang wrote: > > > Seems someone has some anger management issues. As I just stated in > > a previous post, it seems you have issues with me asking valid > > questions. I'm not sure why that is. The long email you rattled off > > with all my questions where quite valid. Your issue with that > is.....? > > > > -----Original Message----- > > From: C F [mailto:shmaltz@gmail.com] > > Sent: Tue 12/20/2005 7:33 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Cc: > > Subject: Re: [Asterisk-Users] SIP Subscriptions > > > > > > > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> > wrote: > > > I don't think the bounties are worth the costs asociated > with > > contracts, legal fees, international boundaries etc. > > > > Right, Avaya is. > > > > This guy is a Fuc*ing retard. > > Douglas, did you see a doc the last few days??????????? > > _______________________________________________ > > --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 > > > > > > <winmail.dat> > > _______________________________________________ > > --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 > > > <winmail.dat> > _______________________________________________ > --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
Like what? I'm being very serious and firmly believe in what I said below. What's your issue with it? I'd really like to know, seriously, because I just can't understand why others don't have a problem with this. -----Original Message----- From: Steve Totaro [mailto:stotaro@totarotechnologies.com] Sent: Tue 12/20/2005 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: Re: [Asterisk-Users] SIP Subscriptions I hope you are just having a breakdown and aren't normally like this. > > Well, I know I will be attacked for saying this, as cursed are those who > say anything bad about Asterisk, but for an application that is > 'enterprise-grade' as Digium tourts, it has several flaws that IMHO > disqualify it as 'enterprise-grade'. I mean, really, c'mon... think about > it... an application that doesn't support the use of a database for common > location/registration information is hardly 'enterprise-grade'. Just off > the top of my head, other disqualifiers would be: > > 1. SIP subscriptions are stored in memory and cleared when you do a > 'reload'. So, if you make any configuration changes and 'reload' you lose > all your BLF lights. People take this stuff for granted and expect it to > work. > > 2. No common SIP registration information. Not even using realtime with > SIP users, which doesn't work, there's no way outside this to share > location info between more than one (ie 'enterprise-grade') Asterisk > systems. > > 3. The 'Dial' application seem to have very limited ability to be able to > determine what SIP response it gets back from a peer. "Not Found", "Busy", > "Moved" etc. I know Asterisk isn't a SIP proxy, but without the ability to > check the SIP message status in a dial, it makes redundancy very very > difficult. Redundancy is normally an important part of 'enterprise-grade'. > Without this, how do you get upstream redundancy? I have something working > right now, but it isn't pretty! > > 4. DNS SRV lookups aren't implemented properly. Another important part of > redundancy and 'enterprise-grade' software. > > Before you all go ripping my head off and defecating down my neck for > saying anything bad about Asterisk (well CF can, I expect that), this is > just my opinion based on real world frustrations with Asterisk. These are > intended to be constructive criticisms. > > Maybe some from Digium will read this email and it will make a 0.001% > contribution towards some of these things being fixed. Oh, and no... I > can't switch to another solution. The decision was made above my head to > go with Asterisk. It's my job to make it do all that 'enterprise-grade' > stuff. > > Doug. > > > -----Original Message----- > From: pdhales@optusnet.com.au [mailto:pdhales@optusnet.com.au] > Sent: Tue 12/20/2005 8:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: Re: [Asterisk-Users] SIP Subscriptions > > > > > C F <shmaltz@gmail.com> wrote: > > > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > Digium needs people like me, if they read this list that is. > They sure > > don't seem to be able to make real-world functionality decisions > on > > their own. > > > > > > > You are so right, how did we do without you the last few > > years??????????? > > By living on baked beans and cabbage. > > PaulH > _______________________________________________ > --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
I don't think anyone will deny that subscriptions have long been a neglected feature. You know... SER might do the trick. :) The default answer for all things SIP that Asterisk does not do. On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote:> > So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose > all your SIP subscriptions. Nice. Basically that means the use of hints and > subscriptions in a production environment is a completely impossible. > Awesome. Considering traditional phone users have come to expect this > functionality, it leaves a lot to be desired as far as Asterisk is > concerned. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/09acb7a2/attachment.htm
>-----Original Message----->From: C F [mailto:shmaltz@gmail.com] >Sent: Tue 12/20/2005 9:58 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Cc: >Subject: Re: Re: [Asterisk-Users] SIP Subscriptions > >Lets get this clear once and for all, you are a Fuc*ing retard, and >his will be the last (I hope) you will hear from me in this post. >>And now to the post. > >On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: >> Well, I know I will be attacked for saying this, as cursed are those who say anything bad about Asterisk, but for an application that is 'enterprise-grade' as Digium tourts, it has several flaws that IMHO disqualify it as 'enterprise-grade'. I mean, >>really, c'mon... think about it... an application that doesn't support the use of a database for common location/registration information is hardly 'enterprise-grade'. Just off the top of my head, > >Like any of the other systems do, lets check which ones? Avaya >Definity? Panasonic? >Why is AstDB not a database? You can query it using asterisk -rx, and >then use your own functions based on the results. In an enterprise >grade installation one doesn't want another point of failure like a >database. A database is only usefull if you want to be able to have >the *same* database for every system, whicn in most asterisk system >this is *not* needed. At the most you want to share the DP which can >be done in other ways. Your not understanding the problem with a common location database. If you have multiple Asterisk boxes, phones may register with any of these Asterisk boxes in a redundant configuration. When that happens and phoneA is registered to Asterisk System 1, it will not be able to find phoneB registered with Asterisk System 2. Asterisk has no way to share this registration information. Each Asterisk box has its own astdb file, but it's not shared and provides no visibility as to the location of phones registered on other Asterisk systems. Asterisk does not provide a mechanism to activate a command upon registration from a SIP phone. That might have been one way to do it, ie 'replicate' somehow the registration to all the other Asterisk systems. Realtime does not support the use of multiple Asterisk boxes accessing the same SIP peers data in the database. It may work sometimes, and not others and therefore it can't be relied upon. DUNDi _might_ do it, but documentation on that is scarce, and the docs I have found are wrong (ie they tell you to use auth=rsa in sip.conf where it's not supported). I've had other people in my department scouring over the scant DUNDi docs for days trying to make heads of tails of it. I'm at a complete loss as to why Digium can't release some docs on it. It's their protocol after all. Why is the same location database not needed for all Asterisk systems? If you want to have them in a redundant fashion, ANY Asterisk system should be able to route calls between any two phones. This requires a central location database (either a real one or a file). Yes, the dial plan can be shared, but I am yet to see a way in which the LOCATION (ie port, ip address etc),that is availability, of a phone can be shared between Asterisk systems. >other disqualifiers would be: >> >> 1. SIP subscriptions are stored in memory and cleared when you do a 'reload'. So, if you make any configuration changes and 'reload' you lose all your BLF lights. People take this stuff for granted and expect it to work. > >I have installed more Asterisk systems than you (yes I'm sure of it, >you are too big of a moron), not ONE of those systems make use of the >hint feature (although I'm looking at it currently), and my clients >are very happy (I use FOP). Tell me how this disqualifies this system >to for enterprise grade, as I have installed in enterprise grade >installs. I don't understand how you can say that. BLF is a standard part of any antiquated PBX system. It's been around forever. People take this functionality for granted and just expect it to be able to work. Why would they want to switch to a VOIP solution if they have to give up functionality??? It's a requirement for our production deployment. >> >> 2. No common SIP registration information. Not even using realtime with SIP users, which doesn't work, there's no way outside this to share location info between more than one (ie 'enterprise-grade') Asterisk systems. >> > >For this #2 question: >A. Why would this be needed? you could use switch to share the DP, or >you could simply have asteriskB just dial extenx on asteriskA which >asteriskA will then dial that SIP >B. You could share it using asterisk -rx "sip show peers" or the like. >C. You could use Avaya. Because the phones may register with ANY Asterisk system. If you force a phone to register with one Asterisk box, it isn't exactly redundant now is it. If you configure the phones to use a specific Asterisk system as primary and another as secondary, it doesn't scale very well. As you add Asterisk boxes, you increase the complexity greatly. I don't believe the switch => command will provide any location data. It doesn't tell you if the user is registered on that other system or not, only that the othe system has a dial plan entry. If the phone isn't registered with that system, you get a failure. Considering that Asterisk doesn't seem to handle Dial() failures very well, I don't believe this could work. >> 3. The 'Dial' application seem to have very limited ability to be able to determine what SIP response it gets back from a peer. "Not Found", "Busy", "Moved" etc. I know Asterisk isn't a SIP proxy, but without the ability to check the SIP message >status in a dial, it makes redundancy very very difficult. Redundancy is normally an important part of 'enterprise-grade'. Without this, how do you get upstream redundancy? I have something working right now, but it isn't pretty! > >There are lots of DP magic in this area that can tell you precisely >whats going on. It's just your inability to RTFM that makes this a >point, I mean realy how can you possibly know enough about Asterisk in >just 14 days????????????? Your ignorants does NOT imply anything about >asterisk. I never said I'd only been working with Asterisk for 14 days. It's been about 2 months. I've been scouring over every conceivable piece of information I have been able to find, and I will tell you that docs on Asterisk are light. Accurate ones are even lighter. I'm yet to see any 'DP magic', or see any intelligent replies on this list to questions I have specifically asked about that. >> >> 4. DNS SRV lookups aren't implemented properly. Another important part of redundancy and 'enterprise-grade' software. > >Work in progress, but not with this atitude, for the mean while use Avaya. Yay. >> >> Before you all go ripping my head off and defecating down my neck for saying anything bad about Asterisk (well CF can, I expect that), this is just my opinion based on real world frustrations with Asterisk. These are intended to be >constructive criticisms. > >You got it all wrong, this is not aimed at you, but it's all aimed at >your local community for not putting you into an institution yet. We >dont' care about you and Asterisk, many of us are using it in >enterprise-grade installs, *YOUR* inability to do so doesn't mean that >asterisk has failed. If you want answers ask, dont bash. >BTW, you are *NOT* anyones real world example with Asterisk. I can point out numerous ocassions where I have asked questions, and not bashed. > >> > Maybe some from Digium will read this email and it will make a 0.001% contribution towards some of these things being fixed. Oh, and no... I can't switch to another solution. The decision was made above my head to go with Asterisk. It's my job to make it do all that 'enterprise-grade' stuff. Oh, I love this last line, does that mean that they will fire you from this asterisk job? I'm looking forward to that. > > Doug. > > > -----Original Message----- > From: pdhales@optusnet.com.au [mailto:pdhales@optusnet.com.au] > Sent: Tue 12/20/2005 8:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: Re: [Asterisk-Users] SIP Subscriptions > > > > > C F <shmaltz@gmail.com> wrote: > > > > On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > Digium needs people like me, if they read this list that is. They sure > > don't seem to be able to make real-world functionality decisions on > > their own. > > > > > > > You are so right, how did we do without you the last few > > years??????????? > > By living on baked beans and cabbage. > > PaulH > _______________________________________________ > --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
Yes. SER has been a saviour here. That was how we managed to get registration info to all the Asterisk boxes, but using SER. Phones register with OpenSER and it forwards() the registrations out. Seems to be working for now, but that solution isn't without it's warts too. -----Original Message----- From: Jon Radon [mailto:jonr800@gmail.com] Sent: Tue 12/20/2005 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Subscriptions I don't think anyone will deny that subscriptions have long been a neglected feature. You know... SER might do the trick. :) The default answer for all things SIP that Asterisk does not do. On 12/20/05, Douglas Garstang <dgarstang@oneeighty.com> wrote: So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4442 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/c2ccd943/attachment.bin
In article <645FEC31A18FE54A8721500CDD55A7B602C152D0@mail.oneeighty.com>, Douglas Garstang <dgarstang@oneeighty.com> wrote:> Good grief. Your perceptions of reality are quite warped. It seems you > have a problem with me asking a lot of valid questions. Why exactly is > that?No-one has a problem with the validity or number of your questions. Many people have a problem with the way you are asking them. Correct the attitude, and you will be much more welcome. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org
Doug, If you stop complaining and listen to what people are saying, you would be able to accomplish your goals. Some of your points have merit, but you are asking for help in all the wrong ways. Please remember, this list is for users, not developers. The user community is quite extensive in their backgrounds. Not all of us are developers or linux experts. There are people on the list that have used Asterisk and installed them in many "enterprise" environments, even though you claim it is not enterprise ready. My point is, instead of annoying everyone and triggering angry replies, you should change your tactic. Right now, you are not getting any useful information and flooding everyone's mail boxes with useless stuff. For example, you already heard that subscriptions in asterisk are work-in-progress. The entire project is work-in-progress. There are stable features and there are new features that are being worked on. If you respect the community, the community will respect you and give you what you need. I have been involved with asterisk for more then 1 year, and have nothing but good to say about people on this list and developers of asterisk.org in general. But, never... never... piss off these people, or you may as well quit your job and go do something else. There are a lot of experience on this list in Asterisk and Data Processing in general (i.e. people smarter then you are ;-). I think one of your mistake is that you trying to depend 100% on Asterisk to do the job. Asterisk is just a growing baby. It is growing fast, but still needs time. As a growing child, it has it challenges/problems and solutions. The solutions may not be very elegant and could only be temporary, but they are solutions to solve immediate business needs. Instead of complaining how inadequate redundancy is with asterisk, you should ask how to architect redundancy with asterisk. I have seen a number of solutions on the list regarding this. There were some that were done purely in asterisk and some were done using SER and Asterisk. Just to prove my point that there are people with solutions out in the community that are willing to help and share their experience, if you ask politely and with respect. Otherwise, you just get angry replies and people calling you nasty names. If you enjoy this, you can continue, but a lot of people will put you in their ignore lists and soon you will be talking to yourself. Ok... that's just my advice to you... you can take or leave it. But I strongly suggest you take it. It will make your job SO MUCH EASIER!!! Again, just my .2c Alex Message: 10 Date: Tue, 20 Dec 2005 20:52:55 -0700 From: "Douglas Garstang" <dgarstang@oneeighty.com> Subject: RE: [Asterisk-Users] SIP Subscriptions To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <645FEC31A18FE54A8721500CDD55A7B602C152D1@mail.oneeighty.com> Content-Type: text/plain; charset="utf-8" Seems someone has some anger management issues. As I just stated in a previous post, it seems you have issues with me asking valid questions. I'm not sure why that is. The long email you rattled off with all my questions where quite valid. Your issue with that is.....?
Hi Ollie. No, Realtime does not support the use of multiple Asterisk systems all accessing SIP users. Digium, including Kevin Fleming, have confirmed this. -----Original Message----- From: Olle E. Johansson [mailto:oej@edvina.net] Sent: Wednesday, December 21, 2005 1:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions Douglas Garstang wrote:> Realtime, as stated by Digium, does not work with sip users. This isn't > related to sip subscriptions. >Realtime certainly works with SIP users. And yes, SIP subscriptions are lost in the current code. In 1.0, the support for subscriptions was very poor so it wasn't used. In 1.2, a few of us *non-Digum* developers enhanced it quite a lot. A community effort, not related to Digium at all. That's why SIP subscriptions is now a useful feature in Asterisk. In 1.3, because we made subscriptions very popular, we have to rewrite the whole subscription system in order to support a large quantity of subscriptions and new features users now want, like shared line apperances and call control. A lot of the work with Asterisk is done by non-Digium developers, in many cases paid for by Asterisk users that wants to enhance Asterisk to better support their business but can't or don't want to develop in house. Digium adds quite a lot of quality control, especially now when they have a team of testers working with the Asterisk Business Edition. I've also started to take Asterisk to the SIP interoperability testing events to further enhance the SIP support in Asterisk. If you want something to change, please do not go out saying "this whole product is crap, since feature yyy does not work as expected and Digium stinks by the way". We are very open to discuss bugs and missing features, as the product is being enhanced. The Asterisk.org developer community is very large, as you can see if you visit the bugtracker and check who is really contributing. Visiting the bug tracker might be a good idea anyway, that's a good place to report your bugs. Good luck with your continued work with Asterisk! Best regards, /Olle _______________________________________________ --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
Ollie. It does now. There's a new chan_sip.c available that I tested last night. Phone reboots, and subsequent new subscriptions do not cause accumulations of sip subscrptions on Asterisk. -----Original Message----- From: Olle E. Johansson [mailto:oej@edvina.net] Sent: Wednesday, December 21, 2005 1:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: SIP Subscriptions Douglas Garstang wrote:> I don't think it's an expiry issue. When a phone reboots before the expiry, it sends a new subscription to Asterisk. Asterisk doesn't remove the old one. If subscriptions where keyed like registrations, the new one would just overwrite the old one.If the phone reboots, it will send a new subscription with a new call ID. This should *not* replace the old subscription with another call ID, since it's a different SIP session. /O _______________________________________________ --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
Ollie. I said it in a previous post. Just to make it clear... :) Realtime does not support having multiple Asterisk systems all accessing a central database for SIP users/registration information. Digium have admitted it doesn't work and have said that it will take the better part of a year to fix. Oh, and on the SIP result codes... how about all of them? Seriously, why not? Or maybe a function to check a SIP result code. -----Original Message----- From: Olle E. Johansson [mailto:oej@edvina.net] Sent: Wednesday, December 21, 2005 1:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Subscriptions> 1. SIP subscriptions are stored in memory and cleared when you do a 'reload'. So, if you make any configuration changes and 'reload' you lose all your BLF lights. People take this stuff for granted and expect it to work.I think we cleared that up in previous postings. Saving the subscriptions in astDB as we save registrations might solve this issue.> > 2. No common SIP registration information. Not even using realtime with SIP users, which doesn't work, there's no way outside this to share location info between more than one (ie 'enterprise-grade') Asterisk systems.Have you checked realtime SIP peers? We do save registration data in the realtime database as well as the astDB. However, if you have NAT between you and the device you need to make sure you send the call from the proper IP.> 3. The 'Dial' application seem to have very limited ability to be able to determine what SIP response it gets back from a peer. "NotFound", "Busy", "Moved" etc. I know Asterisk isn't a SIP proxy, but without the ability to check the SIP message status in a dial, it makes redundancy very very difficult. Redundancy is normally an important part of 'enterprise-grade'. Without this, how do you get upstream redundancy? I have something working right now, but it isn't pretty! This is one of the effects of being a multiprotocol PBX. We have to hide the protocol-layer specific signalling from the dialplan and applications in order to behave in a common way. Can you explain a bit more which SIP error codes that you want to reach and why? Im curious.> 4. DNS SRV lookups aren't implemented properly. Another important part of redundancy and 'enterprise-grade' software. >This is a well known and well documented bug in Asterisk. It's not easy to fix, believe me, I've tried for a long time. The new DNS manager is a way forward and many of the core developers are trying to build a foundation to solve this issue once and for all. /O _______________________________________________ --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
Douglas Garstang wrote:> Hi Ollie. No, Realtime does not support the use of multiple Asterisk systems all accessing SIP users. Digium, including Kevin Fleming, have confirmed this.It has happened that Kevin has been wrong about the SIP channel, you know. Or you might have asked the wrong question. Are you talking about SIP users (type=user) that call in to your Asterisk? Or Peers that we place calls to? We do handle both users and peers in realtime. We do save peer registration data in realtime. Both has been proven to be working in situations with multiple Asterisk systems. /O