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

[Eric "ManxPower" Wieling]

Leif Madsen wrote:
> 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.

I suspect there are many more people using 1.2 than 1.4.  You live on 
the cutting edge of Asterisk development.  Most people want a PBX that 
Just Works and for many people 1.2 is that PBX.