[Asterisk-Dev] chan_exosip2

[Kevin Hanson]

Olle E. Johansson wrote:

>>I am referring to extension/device presence.   From Josh's voip-forum.com:
>>    
>>
>I didn't that Josh had kidnapped my site and domain ;-)
>  
>
Sorry.  Earlier in the thread you said "Josh has been working on some 
changes (covered on www.voip-forum.com)".  I wrongly assumed that it was 
his site :-[ .

>  
>
>>"Asterisk 1.2 will have support for SIMPLE notifications of device
>>presence. We will show if a phone is on a call, if it's ringing or if
>>it's off line (not registered)."
>>
>>From what I understand, this only works if call-limit is set to 1 for
>>the devices in sip.conf.  This has the nasty side affect of not allowing
>>that device to have call waiting, attended transfer etc.
>>    
>>
>This only works if you have a call limit, but you can set it to 10 if you
>want to. However, we will not force busy if we are still able to call
>the phone.
>  
>
Ok.  If I set call-limit to 10 on device A, will a subscribing device 
(B) show device A "in use" when less than 10 calls are in progress?  Do 
"busy" and "in use" mean the same thing?

>  
>
>>Is removing the call-limit requirement require a new architecture?  I
>>would like to see this happen and am trying to understand the big
>>picture effort-wise so I can post an appropriate bounty, do it myself,
>>hire someone, etc.
>>    
>>
>Since I developed this part of chan_sip, I am interested in what it is
>that does not work if you set the call limit to 10. I might be missing
>something here that we need to fix. Maybe a force-busy-on-call-limit=1
>setting that does not enforce the actual call limit that could be set to
>a higher value.
>
>/O
>
>  
>
I guess I need to try with call-limit set to a value greater than 1 and 
see what happens.  Currently I don't have call-limit at all in sip.conf 
because I thought it had to be 1 for device presence to work.

I have a customer with a receptionist that has a Polycom 601 w/ side 
car.  They are running Asterisk 1.2 rc 2.  I have all the other 
extensions setup as buddies on the receptionist phone.  When the phone 
first boots up it appears that presence is working great.  An associate 
is on the phone and the associated button and light on the receptionist 
phone indicate "in use". 

However, after a period of time presence stops working, or only works on 
some of the subscribed extensions (it's not consistent which ones are 
working at any particular time).  [One note...the receptionist phone 
always correctly shows that the other extensions are "on line" (i.e. 
registered w/ asterisk), so "on line" state works.]  It just doesn't 
always correctly show the other device states I need...ringing or in 
use.  As a work-around they are using Flash Operator Panel to see device 
state.

I get a lot of:

Incoming call: Got SIP response 500 "Internal Server Error" back from 
192.168.0.124

I have just been assuming this was related to incomplete support of 
presence and haven't looked at sip traces.  Now I'm thinking these 
errors aren't related to presence and need to see what's causing them.

Thanks for your help.  I am actively monitoring the development of this 
feature of asterisk and would like to help out any way I can.  Every 
customer proposal wants this feature.

Cheers,
Kevin