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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
