Michael, Thanks for the feedback, I'll check it out and let you know...
Paul ----- Original Message ----- From: "Michael Ulitskiy" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 30, 2003 6:17 AM Subject: Re: [Asterisk-Users] Already on the phone? > Paul, > > Thanks. Unfortunately your patch doesn't work reliably. > I specified incominiglimit=1 and placed a call to extension 12125551111. > Now I have: > 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 1 1 0 N/A > 12125550028 0 N/A 0 N/A > 12125550014 0 N/A 0 N/A > > 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 > > On Tuesday 28 October 2003 07:20 pm, Paul Liew wrote: > > Michael, > > > > I've added a patch a week ago on to bugtracker to fix this - feel free to > > try it and let me know > > > > http://bugs.digium.com/bug_view_page.php?bug_id=0000408 > > > > Paul > > ----- Original Message ----- > > From: "Michael Ulitskiy" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Wednesday, October 29, 2003 10:31 AM > > Subject: [Asterisk-Users] Already on the phone? > > > > > > > Hi, > > > > > > I'm wondering if there's a way within a dialplan or AGI to find out > > > if an extension (SIP client) is already in use and the > > > person is already on the phone? > > > By default the channel is assumed available and callwaiting tone > > > is transmitted to the called extension. AFAIK there's no way to turn > > > off callwaiting from within the dialplan. > > > I need to avoid the callwaiting behavior in some cases and pass the > > > call to another extension if called extension is already in use. Is this > > > possible with asterisk? > > > I've tried chanisavail application, but since callwaiting is enabled it > > > always returns true. > > > 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 > > > > _______________________________________________ > 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
