asterisk users - Mar 2004 - Asterisk interoperability w/ new 64bit processors & SIP express router

HEY! I'm doing research and testing for my Thesis on a prototype SIP PBX for
a
facility of 20-30 users. (T100P / Atlas 550series / Cisco Routers &
switches)

A couple of concerns that have come up are:

1. Has there been any known issues concerning asterisk with the new 64-bit
processors? 

2. Asterisk is SIP compatible, but to my understanding it doesn't have
support
for SIP registrar, proxy or redirect server. Please correct me if wrong.
I've
yet to make a decision on which Sip server to use, so any ideas would be nice.
SER was on my mind but the question is whether I can integrate it directly on
the same linux server running Asterisk without complications or does it need to
be separate. 

Comments, ideas and experiences would be greatly appreciated. These were a
couple of subjects concerning me and couldn't seem to find answers to.

By the way, I will be Documenting all testing and issues + much more on my
homepages too. It will include a lot on SIP areas and ofcourse Asterisk* ! 


I'll make those available in the near future. 

Thanks ahead!

t: Mike

Hi,

The messages produced by asterisk console, in vvvvvvv mode,
what are the numbers after the brackets?

in this example, /4 and /5

=> Releasing IAX2[stig@stig]/4 and IAX2[ulf]/5

Are these session numbers or?
Are they reused?

When the first call comes after asterisk is restarted, they begin at  /1
but 8 hours later, a new single call can have /4

I'm investigating why some calls do not go through to a Firefly client
(IAX2)
after the client has been busy. I'm suspecting som kind of zombie
sessions...

anyone? 
Any ideas?

/Stig