[Asterisk-Users] Fw: anybody has SIP realtime working ?

[Douglas Garstang]

><rant>
>*SCREAMS*
>
>Tell me, why is someone as sarcastic as you and with such a caustic
>attitude towards an OPEN SOURCE project that is maintained and fixed
>primarily by people on their OWN TIME even looking at this project.
I'm not looking at this project. It's already been chosen. 
 
>
>Day in and day out, you find something you don't like, and spend a week
>bitching to the asterisk-users list about how asterisk isn't worth this
>and isn't worth that.  If you don't like it, don't use it.  Go with Cisco,
>where everything is server centric, or go grab a Nortel system, so you
>won't have to worry about registrations.
Have you ever considered that maybe this is because there's plenty of things not to like? I don't know how high your standards are, but they must not be as high as mine. I don't seem to recollect any specific situations where I have said asterisk isn't worth this or that. I do recollect many specific occassions where I have asked questions because Asterisk either didn't work the way I expected, didn't work the way the docs implied, or didn't work in a Enterprise Class' fashion that Digium states it as. You try spending all trying to get something apparently simple to work, and see how much it frustrates you sometime. Your statement about me 'worrying about registrations' tends to emphasize the fact that your standards are indeed not very high. People using 'Enterprise Class' software expect it to work in a 'Enterprise Class' fashion. That means that your service doesn't go down for N number of minutes while you wait for a phone to reregister, in order to get around some Asterisk HA limitation.
 
>
>Unless I'm totally off base on this, SIP is totally phone centric, with a
>proxy to let phones know where each other is.  Hell, you can generally
>drop a SIP phone on a desk, and just dial IP's and it wouldn't give a
>damn.  Asterisk is just that kind of device, it doesn't have to stay in
>the phone call to know what's going on, and if you program it right, if
>you lose a phone call during conversations, you won't lose those
>conversations, just the cdr's that go with them.
It does have to remain in the call path if you want DTMF services such as call recording, which we do. Not sure where you where going with the rest of that paragraph.

>
>Yes, you have to prune your database entries, or just wait a few minutes
>for the phone to re-register if you want information from a database
>backend to work.  That's how it works, if you don't like it, rewrite it,
>or give some CONSTRUCTIVE criticism to the deveopers.  In here, we use
>Asterisk because we like how it works, and it works great for us.  I'm in
>the middle of a huge rollout because we're extremely happy with the
>system.  I'm sorry you're not, and that's all I can really say unless
>you're quite a bit more cordial about your responses and requests.
Telecom customers of an 'Enterprise Class' pbx solution don't expect to have to wait minutes, with no ability to receive calls, for updates to happen to their account. Now, I would give constructive criticisim if I knew how it worked. I spend all my time just trying to understand that, between bad documentation, and downright hostile attitudes towards anything bad said about Asterisk by the likes of yourself.  I've spent the last several hours dicking around with Queue functionality and it's behaving not at all like the docs say. How can I make a constructive criticism about something I don't understane the function of? How can I make a constructive criticism about how realtime works when it isn't properly documented anywhere? If there where some hard and fast facts, I could counter those and constructively say why I didn't like it, but such a thing doesn't exist so I'm stuck with bits and pieces here and there and trying to piece it's function together from my own observations.

>
>If I'm totally off base, ignore me, but I'm tired of the constant remarks
>and rude comments about asterisk and it's developers.  Those of us that
>have been here for a while are here because collectively, we know what it
>takes to learn asterisk.  Hell, some of us have even helped out a little
>with the development.
Which rude remarks towards developers are you specifically referring to? I don't seem to recollect any. I have actually received a number of personal emails from people who completely understand where I am coming from and are just as frustrated as I, both my Asterisk's limitations (specifically HA), and the hostile attitudes towards people who even consider questioning Asterisk's ability to save the world. 

</rant>

Aaron

On Thu, 23 Mar 2006, Douglas Garstang wrote:

> Please don't tell me what I think you are. Are you saying that to change a configuration setting for the phone I have to remove it as a peer, and then wait for it to re-register? Are you serious???
>
> Doug.
>
>> -----Original Message-----
>> From: Aaron Daniel [mailto:amdtech at shsu.edu]
>> Sent: Thursday, March 23, 2006 2:54 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: RE: [Asterisk-Users] Fw: anybody has SIP realtime working ?
>>
>>
>> Maybe if I repeat myself like 18 times.
>>
>> sip prune realtime <exten>
>>
>> USE THIS COMMAND... it clears the settings on a single phone,
>> leaving all
>> the others untouched.  We do it all the time when we
>> reconfigure a phone.
>>
>> Aaron :)
>>
>> On Thu, 23 Mar 2006, Douglas Garstang wrote:
>>
>>> Ok Andrew. Here's one for you... I just changed qualify
>> from yes to no in the database... a 'sip show peers' still
>> showed Asterisk as qualifying the users... I had to do a
>> reload to get to accept the change to the database.
>>>
>>>
>>>> -----Original Message-----
>>>> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk at benshaw.com]
>>>> Sent: Wednesday, March 22, 2006 10:46 AM
>>>> To: asterisk-users at lists.digium.com
>>>> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime
>> working ?
>>>>
>>>>
>>>> On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
>>>>> First thing that comes to mind, what if we decided to
>>>> change a non user
>>>>> setting in sip.conf?
>>>>
>>>> You're reaching.  You said you NEED to reload all the time,
>>>> that this is a
>>>> MAJOR issue, a deal breaker.  So surely you must have
>>>> experienced this
>>>> downtime to be so sensitive to it.  What did you do on your
>>>> PRODUCTION system
>>>> that required constant reloads to cause the current behavior
>>>> to be such a big
>>>> problem?
>>>>
>>>> Honestly; if you're changing a non-user setting in sip.conf
>>>> you're going to do
>>>> that very, very infrequently, and you'd do it during a low
>>>> volume time.
>>>>
>>>> You said this is a major problem.  I'm calling you on it.
>>>> I'm interested in
>>>> making Asterisk robust and highly-available too, but I'm not
>>>> making up
>>>> scenarios in order to launch complaints and verbal assaults
>>>> against the
>>>> project in order to feed my inflated ego and try to get
>>>> things done "my way."
>>>>
>>>> If you have a specific problem, let's hear it.
>>>>
>>>> -A.
>>>> _______________________________________________
>>>> --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
>>>
>>
>> --
>> Aaron Daniel
>> Computer Systems Technician
>> Sam Houston State University
>> amdtech at shsu.edu
>> (936) 294-4198
>> _______________________________________________
>> --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
>

--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
amdtech at shsu.edu
(936) 294-4198
_______________________________________________
--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