[Asterisk-Users] extensions problem (SIP)

[Jon Lawrence]

On Monday 15 March 2004 20:35, Walker Haddock wrote:
> On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
> > The incominglimit limits how many simultaneous calls a UA may place to
> > Asterisk.
>
> I'm pretty sure that the incominglimit specifies how many calls that * can
> send to the SIP device.  If you set incominglimit=1 and then do a SIP SHOW
> INUSE from the *CLI then you will see the limit set.  The behavior of *
> then will consider the device busy if there is a call in progress and the
> inuse count is incremented.
>
> Paul Lieu did some work on this a few months ago and I've been using it on
> my Cisco 7960 and Grandstream BT-102 phones.

The interface to my handytone is identical to a BT-102 so it may also work 
with the handytone :). Where did you specify incominglimit=1 - is it in the 
sip.conf for that UA ?

Jon