[Asterisk-Users] SIP Subscriptions

[C F]

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 at 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.