If the phone has two lines on it, you can be creative and set them up differently. (one for incoming, no limit. one for outgoing, limited to 1)
PaulH ----- Original Message ----- From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Sent: Monday, November 07, 2005 3:35 AM Subject: RE: [Asterisk-Users] limiting incloming call on sip phones to 1 > Hi Kebin., > > Thx for your comments, their exactly what I read. Problem comes when you > want to be able to make any number of incoming calls (calls from the phone > out) but limit the number of outgoing calls (calls from asterisk to the > phone). > > :( > > |-----Original Message----- > |From: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] On Behalf Of > |Kevin Hanson > |Sent: Sunday, November 06, 2005 9:09 AM > |To: Asterisk Users Mailing List - Non-Commercial Discussion > |Subject: Re: [Asterisk-Users] limiting incloming call on sip > |phones to 1 > | > |Anton Krall wrote: > | > |>Hey Guys! > |> > |>I know sip hpones can be configured to disable call waiting > |but this is > |>for all call appearances. I was wondering if there is a way to limit > |>outgoing calls (asterisk -> phone) to a sip phone (techonology) to 1? > |> > |>Is there any other way of doing this without groups or such? Any kind > |>of settings on sip.conf for this? > |> > |> > |>_______________________________________________ > |> > |> > |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / > |CVS head) in sip.conf for that extension. > | > |These limits are named from asterisk's perspective. > |incominglimit is calls coming in to asterisk, so it would > |limit calls from the sip phone to asterisk, but not from > |asterisk to the phone. outgoinglimit (1.0.x) doesn't work > |from what I've read. > | > |call-limit is both directions. It may be what you need. > |However, you won't be able to do an attended transfer. Blind > |transfer might work, but I haven't tried it. > | > |quote from previous thread from Olle Johansson: > | > |"incominglimit is replaced by call-limit. Please read sip.conf.sample. > | > |Outgoinglimit has not worked for ages, so we removed it. One > |limit works for both incoming and outgoing calls now." > | > |_______________________________________________ > |--Bandwidth and Colocation sponsored by Easynews.com -- > | > |Asterisk-Users mailing list > |[email protected] > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
