[Asterisk-Users] New Asterisk bounty: SIP simultaneous

Sun Jul 11 02:15:11 MST 2004

I explained him a sample need.
I don't think asterisk does whatever i want in sip. It is an good PBX.

Please help me to understand. Anywhere am I wrong ? Or as you say is that
SIP feature is written?

-Kannaiyan.

----- Original Message -----
From: "usedcanon" <usedcanon at yahoo.co.uk>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, July 11, 2004 10:02 AM
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

> I was going to keep out of this (was interesting to read, as I have dealt
> with simmillar situation) however I would like to add just this one
commnet.
>
> Try to better understand asterisk than to throw about your money. What you
> want to do is perfectly possible with asterisk there is no need to add a
new
> confusing feature.
>
> As for your bounty, donate it to the wiki ! :-)
>
> Umar.
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Kannaiyan
Natesan
> Sent: 11 July 2004 09:51
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
>
>
> 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
> >
> > __________________________________________________
> > Do You Yahoo!?
> > http://bb.yahoo.co.jp/
> >
> > _______________________________________________
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> >
>
>
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