Re: [Asterisk-Users] Preventing incoming calls from ringing SIP lines
On Dec 5, 2005, at 9:02 PM, James B. MacLean wrote: Paul Redstone wrote: Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. Just a newbie response, but what about the incominglimit= option in your /etc/asterisk/sip.conf? I think incominglimit is deprecated. Doesn't x-Lite have a setting that lets you disable call waiting? That's how we handle this problem with Cisco 79XX. Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 "Intelligent technology solutions for small businesses." ___ --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
Re: [Asterisk-Users] Preventing incoming calls from ringing SIP lines
Paul Redstone wrote: Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. Just a newbie response, but what about the incominglimit= option in your /etc/asterisk/sip.conf? JES smime.p7s Description: S/MIME Cryptographic Signature ___ --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
[Asterisk-Users] Preventing incoming calls from ringing SIP lines
Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. In 1.0 the SetGroup and CheckGroup commands could do this but you have to build it into all parts of the dial plan. In 1.2 these do not exist and the Set(Group type commands with GotoIf are supposed to be used. But I still have not seen anywhere a full example of this. There is the call-limit setting in SIP - beautiful, works at the SIP level so easier than the dial plan. BUT with this you cannot do attended or blind transfers - not sensible. This must be a very common requirement, certainly is judging from the posts but in hours of searching I have not see the sort of complete solution which looks feasible. Thanks and sorry if I've missed it. Alternatively I'd be happy to use single line SIP softphones but cannot find one which feels good. TIA Paul R ___ --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
