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

[Aaron Daniel]

<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.

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.

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.

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.

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.

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