In environments where users have their hard and soft phones... How do you glue everything together?
|-----Original Message----- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Martes, 21 de Junio de 2005 07:39 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice | |> I would like to hear tips and tricks on extention config best |> practices, for example, naming, etc. and most of all, how to |deal with |> extention that have a full time hardphone configured and |assigned and |> then a softphone connecting to the same extention, for example, one |> employee has its hardphone on the office but sometimes when |he travel, |> he uses his softphone to work with, what happens when two |phones have |> the same user id and connect to the same asterisk? How are calls |> routed or how to handle this kind of scenarios. | |In general terms and without being able to see how the |extension is defined in sip.conf, the last phone to register |with * will get the call. | |Assuming both the hard and soft phones register every hour, it |is entirely possible the hard phone will get the call for the |first 30 minutes and the soft phone for the next 30 minutes. | | |_______________________________________________ |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 | _______________________________________________ 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
