Michael, A couple of things - having a quick look at the app_ChanIsAvail code - it seems that it is designed for Zap devices, so using them on any SIP phones would not provide the expected result. Secondly, which SIP phone are you using, I can't put calls on hold and make further calls without parking them. In either case, I suspect the call has been palmed off to asterisk, otherwise you wouldn't be able to make further outgoing calls (the incoming limit would block it). The inuse limit would apply while you are actually in a call. Does it work when you take the original call back off hold ??
I think having the ability to change the incominglimit from the dialplan might be a good idea, but I think prior to any discussion on that, this patch would have to be proven to work reliably and if approved by Digium - put into the CVS. Paul > I put it on hold and placed a few other calls. Then I see: > pbx1*CLI> sip show inuse > Username incoming Limit outgoing Limit > 12125550011 0 N/A 0 N/A > 12125559999 0 N/A 0 N/A > 12125552222 0 N/A 0 N/A > 12125550029 0 N/A 0 N/A > 12125550012 0 N/A 0 N/A > 12125551111 0 1 0 N/A > 12125550028 0 N/A 0 N/A > 12125550014 0 N/A 0 N/A > > So it looses status of existing call somehow. Now callwaiting is > there again. It seems that the status is lost after calling chanisavail > application, although I'm not sure about that. > Also if I can make a suggestion it would be great not to have > incominglimit set statically per client, but have an application > to change it from dialplan (have no idea how hard it is to implement). > If there are other ways to check if the line is already in use or > turn on/off callwaiting on SIP clients, that would also be very > nice and desirable feature. > Thanks. > > Michael _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users