[Asterisk-Users] New Asterisk bounty: SIP simultaneous

[Kannaiyan Natesan]

I accept your views.

I have a specific requirements, can you help to attain the same.
In our business we have 25 employees handling customer service.

I want to add or remove employees in the customer service so does the
devices connected to it.
I don't want to make any changes in the asterisk, and all I need is to plug
in the VoIP Phone and start handling the customer service. I would like to
do for as many employees as I want without any problems.

Can you think of a better solution?

-Kannaiyan.

----- Original Message -----
From: "Sunrise Ltd" <stsltdtyo at yahoo.co.jp>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, July 11, 2004 9:15 AM
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous


> >When I call a SIP user, the phone should ring in more
> than one
> >extentions. Also more than one phone should be able to
> register with
> >asterisk. Right now it is not the case.
>
> There is no issue here. You seem to be confused, that's
> all.
>
> A SIP account is a SIP account and an extension is an
> extension. You can assign an extension to an account (or
> to multiple accounts) and the tool for that is the dial
> command.
>
> However, there is no implicit assignment between an
> extension and an account and that is good so. This should
> not be changed because it would harm Asterisk's
> flexibility and manageability.
>
>
> >This type of situations might be needed in call centres.
> >
> >Called 12345
> >            |-----------(12345) Ringing
> >            |-----------(12345) Ringing
> >            |-----------(12345) Ringing
>
> As I said, you are confusing extensions with accounts. The
> first "12345" is an extension, the three "(12345)"s are
> accounts. Those are different layers, don't mix them up.
>
> You should always be able to distinguish between devices,
> even if they are assigned the same phone number. In fact,
> in a call centre you'd be using a call queue. It would be
> rather nonsensical for a call queue management to have to
> distinguish between multiple identical agents.
>
> Therefore, setting up multiple devices with the same
> account credentials is not a good idea, especially not in
> a call centre. Each device and each agent should have
> their own unique account credentials and assigning
> extensions to them should always be done through the
> dialplan and only the dialplan.
>
> Asterisk has been designed this way. It is a good design.
> It should NOT be changed nor undermined.
>
> You may want to do something like this ...
>
> [GLOBALS]
>
> A-GROUP => SIP/2001 & SIP2002 & SIP/2003
>
> B-BROUP => SIP/jdoe & SIP/dflint & SIP/bsmith
>
> ...
>
>
> [Support]
>
> exten => 12345,1,Dial(${A-GROUP},30,r)
> ...
>
> exten => 54321,1,Dial(${B-GROUP},30,r)
> ...
>
>
> There is of course an issue when you want to let different
> phones start ringing at different times, for example, the
> first phone is supposed to start ringing immediately and
> the other two are only to join in if the first phone
> hasn't been picked up in 10 seconds, like so
>
> exten => 12345,1,Dial(${JDOE},10,r)
> exten => 12345,2,Dial(${JDOE}&{DFLINT}&${BSMITH},20,r)
>
> This works but if JDOE was to pick up right between those
> two dial commands, then it will have been too late for the
> first and JDOE will be "on the phone" for the second dial
> command, so there is some room for improvement. A bounty
> might better be spent on solving this little problem.
>
> Also, Asterisk supports call groups and pickup groups.
> Indeed, there have been some bugs with those features and
> I am not sure if they have have been fixed, but if they
> haven't, then it would again make more sense to put the
> bounty on fixing those rather than creating an ugly
> workaround.
>
>
> >I feel this is a great feature
>
> I don't and if you spent some more time with Asterisk and
> immerse its philosophy, then you'll very likely change
> your mind.
>
> >in other SIP proxy server this can be done easily
>
> Asterisk is not a SIP proxy. It's a telephone exchange.
>
> >i mean its default 1 or more phone could be registered
> >at 1 number (12345) and resulting same effect
>
> A phone does not register at a number. It registers at an
> account to which Asterisk can assign one or more numbers.
> This makes perfect sense and it is a far more flexible and
> better design.
>
> SIP proxies' auto assignment of extensions to SIP
> usernames is a serious limitation, not an advantage.
>
>
> The only situation where one might want to consider
> supporting multiple concurrent logins on the same account
> is for public VoIP service providers where end users might
> have a SIP phone on their desk and use a softphone on
> their notebook when they are traveling.
>
> But here again, it is more likely to be a disadvantage.
> Consider the following situation ...
>
> 1) Incoming call to 12345
>
> 2) both deskphone 12345 and road warrior's notebook 12345
> ring
>
> 3) Secretary of Mr. 12345 picks up before he himself is
> able to do so
>
> 4) Caller asks for Mr.12345 but secretary has no way of
> trying to transfer the call
>
> OTOH, Asterisk handles this situation much better ...
>
> 1) Incoming call to extension 12345
>
> 2) Dial command determines to ring both deskphone and road
> warrior's notebook which are on different extensions
>
> 3) Secretary of Mr. Road Warrior picks up before he
> himself is able to do so
>
> 4) Caller asks for Mr. Road Warrior, secretary transfers
> to internal extension of road warrior notebook's softphone
>
>
> I am sorry but your bounty doesn't seem to make sense. It
> looks more like one of those "Wanted: problem for given
> solution" cases.
>
> rgds
> benjk
>
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