[asterisk-dev] Status of SIP call-limit in 1.2 branch

[Leif Madsen]

On 10-Oct-07, at 12:02 PM, Eric ManxPower Wieling wrote:

> Leif Madsen wrote:
>
>> I believe you need to use call-limit in sip.conf if you want presence
>> information, but I don't use that, so I can't tell you any more about
>> it :)
>
> You only need it in 1.4 for presence.  In 1.2 you do not need it for
> presence.

Right. I guess I've not used 1.2 in over a year, so I don't even  
think about how it works anymore. I just assume 1.4.

Leif.