Could you just configure the extention to be a ring group instead of an actual extention, or ring queue.. then have his phone/laptop log in whenever he's at the office/coffee shop?
I know AMP has the functionality, but I haven't gone behind the scenes and looked at the sip.conf or extensions.conf to see what the script or macro is doing in a ring group/queue. Daniel -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Tuesday, June 21, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Extension Configuration Best Practice Ok, so how are you guys coping with scenarios like this? Managers working in the office during the day or mid day and then in the afternoon, working remotely using their laptops? |-----Original Message----- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 08:20 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice | |Rich is indeed correct, Asterisk does not yet support multiple |registrations for a single peer entry. Thus when you register |the previous registration is discarded and the new one is |used. Thus like he said, the last one that registered gets the call. | |- Joshua Colp. | | |On 6/21/05 9:39 AM, "Rich Adamson" <[EMAIL PROTECTED]> wrote: | |>> 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 | _______________________________________________ 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
