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. 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. My "best" thought so far is a Macro to check each SIP entry that has CWI disabled, using SetGroup(), and removing it from the dial string if it is in use... Any better suggestions out there? Thanks, Steve _______________________________________________ --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