[asterisk-users] CWI, call-limit and incominglimit

[Olle E Johansson]

23 feb 2007 kl. 12.42 skrev Steve Davies:

> Hi,
>
> In older versions of asterisk I used to be able to use
> "incominglimit=1" to effectively disable call waiting on a specific
> SIP channel (Where broken phones do not allow this on the handset
> itself)

>
> In 1.2.x this became "call-limit=1", but this prevents the phone from
> opening a 2nd line in order to transfer a call using attended
> transfer. The WiKi suggests using SetGroup() etc, but this does not
> cater for the case where you are Dialling several different phones
> simultaneously.
>
You can still set one call-limit for the user and another for the
peer. The peer call-limit would be used to prevent call waiting
and the user limit could be set to a reasonable level so the phone
can do transfers.

> I _could_ dial a whole bunch of Local channels, each of which checked
> for an extension usage count, but the additional load and complexity
> in the dialplan seems a bit over-the-top to me, especially when there
> used to be a one-line solution to this.
>
> I also considered separate user and peer sections in sip.conf, but the
> hosts are dynamic, and there is no way to link the IP address of the
> peer to the user.
>
Why is that an issue? The user authenticates on the incoming call,
no IP address is needed since the auth is done on the From: header.


/O