[Asterisk-Users] Extension Configuration Best Practice

[Anton Krall]

I guess I would need to do something like that and mix with dialing 2
extension at the same time with dial(ext1&exte2)

Seems the easier way to do it for now. 

|-----Original Message-----
|From: asterisk-users-bounces at lists.digium.com 
|[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
|Rich Adamson
|Sent: Martes, 21 de Junio de 2005 10:25 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Extension Configuration Best Practice
|
|One method is to give each hard and soft phone their own 
|extension numbers. Then create a 'call forwarding' approach 
|that essentially says when someone dials x1234, ring 1235 
|instead. I believe there are a couple of asterisk-based call 
|forwarding approaches shown in the wiki.
|
|One such way is to have your user dial a predetermined 
|extension (eg, 4123) and the code within that extension 
|definition does a dbput of some value (eg, true, 1, or 
|whatever). Then when someone calls x1234, test the value using 
|dbget to see if the call should be forwarded. That's all 
|hard-coded logic, but the wiki has some macros to do that same 
|kind of thing.
|
|------------------------
|> In environments where users have their hard and soft 
|phones... How do 
|> you glue everything together?
|> 
|> |-----Original Message-----
|> |From: asterisk-users-bounces at lists.digium.com
|> |[mailto:asterisk-users-bounces at lists.digium.com] 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
|> |Asterisk-Users at lists.digium.com
|> |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
|> Asterisk-Users at lists.digium.com
|> http://lists.digium.com/mailman/listinfo/asterisk-users
|> To UNSUBSCRIBE or update options visit:
|>    http://lists.digium.com/mailman/listinfo/asterisk-users
|> 
|
|---------------End of Original Message-----------------
|
|
|_______________________________________________
|Asterisk-Users mailing list
|Asterisk-Users at lists.digium.com
|http://lists.digium.com/mailman/listinfo/asterisk-users
|To UNSUBSCRIBE or update options visit:
|   http://lists.digium.com/mailman/listinfo/asterisk-users
|