asterisk users - Dec 2005 - Realtime Multiple Asterisk boxes and rtcachefriends MWI

I am working an a multiple box asterisk solution.   I need phones to be able to
login to multiple asterisk servers.  I need Phone A to be able to register to
switch A and call Phone B that is registered to switch B.

 

With rtcachfriends=no  this can be done, However I then loss MWI and "sip
show peers" plus if a Phone becomes unreachable the phone I get dead air
until the dial timeout reached.

 

With rtcachfriends=yes  I get MWI & "Sip show peers", However I
cannot call phones that register to a different switch.

 

My current working solution is to have rtcachfriends=yes.  Place the call via
sip if dialstatus= chanunavaliable

I then route the call to the other switch via an IAX trunk.  Everything works
but I don't have a true load balance soltuion. Plus it really only works for
2 boxes. It get out of hand when I add more..

 

I have tried using AGI and dialing the "full contact" found in the SIP
realtime table. It works if the phone is active, but if the phone is no active I
get dead air until the dial timeout is reached. This will not work as I cannot
have 12 sec of dead air.    So is there a way know the status of a SIP UA?  It
is it in the SIP realtime data?   I looked at regseconds but it does not seem to
be it because I can have a UA that is unreachable and the regseconds are not
expired.

 

Could realtime be altered to add a status filed to the SIP realtime table?   

 

Or is there a asterisk configuration option that I missed?       

 

This is my first post so please forgive me if I posted this in the wrong list.

 

 

Many thanks!

Doug Gillespie

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I'd be very interested in hearing more about this
as I am in need of a similar installation. Anyone
have a hint?

  Kristian

On Thu, Dec 29, 2005 at 02:14:29PM -0500, Asterisk
wrote:
> I am working an a multiple box asterisk solution.   I need phones to be
able to login to multiple asterisk servers.  I need Phone A to be able to
register to switch A and call Phone B that is registered to switch B.
> 
>  
> 
> With rtcachfriends=no  this can be done, However I then loss MWI and
"sip show peers" plus if a Phone becomes unreachable the phone I get
dead air until the dial timeout reached.
> 
>  
> 
> With rtcachfriends=yes  I get MWI & "Sip show peers", However
I cannot call phones that register to a different switch.
> 
>  
> 
> My current working solution is to have rtcachfriends=yes.  Place the call
via sip if dialstatus= chanunavaliable
> 
> I then route the call to the other switch via an IAX trunk.  Everything
works but I don't have a true load balance soltuion. Plus it really only
works for 2 boxes. It get out of hand when I add more..
> 
>  
> 
> I have tried using AGI and dialing the "full contact" found in
the SIP realtime table. It works if the phone is active, but if the phone is no
active I get dead air until the dial timeout is reached. This will not work as I
cannot have 12 sec of dead air.    So is there a way know the status of a SIP
UA?  It is it in the SIP realtime data?   I looked at regseconds but it does not
seem to be it because I can have a UA that is unreachable and the regseconds are
not expired.
> 
>  
> 
> Could realtime be altered to add a status filed to the SIP realtime table?
> 
>  
> 
> Or is there a asterisk configuration option that I missed?       
> 
>  
> 
> This is my first post so please forgive me if I posted this in the wrong
list.
> 
>  
> 
>  
> 
> Many thanks!
> 
> Doug Gillespie
> 
> _______________________________________________
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-- 
Kristian Larsson, Net At Once AB
Email: kristian@netatonce.se
Phone: +46 470 592717
Cell: +46 704 910401

I am having a problem with sip in asterisk 1.2.1 & 1.2.8 . I have an account
setup with a sip provider. The inbound call is coming from a SIP proxy, the call
is setup (I have audio) and then drops down after 15sec.

What I see in sip traces is that the sip proxy is sending "200 ok"
asterisk is responding with a "ACK" however the ACK is send to the to
a different host then the one that sent the "200 ok". After the remote
proxy retransmits a few times and receives no ACK it sends a BYE.

I tried loading trunk, (from a few days ago) and this problem appears to be
fixed as it works just fine. The only problem is this is a production system and
I do not feel ok running trunk. What I would like to do is just load the patch
for this problem then wait for 1.4 Release. I have been reading the SVN log for
chan_sip, however I am unable to identify the problem.

Anyone know what fix solved this problem?   Any tips to finding it in SVN?

Thanks.

 

 

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