Paul, I'm using Cisco 7960 phones. I did some more testing and it looks like using chanisavail with SIP channel causes it loose inuse status. I've removed chanisavail application from dialplan and now I cannot reproduce the problem whether the call is on hold or not. So you patch is probably fine. I'm impatiently waiting for incominglimit application ;-) Thanks.
Michael On Wednesday 29 October 2003 05:28 pm, Paul Liew wrote: > 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 > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
