[Asterisk-Users] Extension Configuration Best Practice

[Goolsby, Daniel S (Daniel)]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com 
|[mailto:asterisk-users-bounces at lists.digium.com] 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" <radamson at routers.com> 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.
|> 
|> 
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