[Asterisk-Users] SIP Subscriptions

[Olle E Johansson]

Douglas Garstang wrote:
> Ollie.
It's "Olle" :-)
> 
> 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.

There are more developers out here than those that work for Digium.
And by the way, SIP users does *not* register in Asterisk.

> 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. 
> 
Can you please explain a bit more how that would help you?

/Olle

> 
> -----Original Message-----
> From: Olle E. Johansson [mailto:oej at 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. "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!
> 
> 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
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