[Asterisk-Users] New Asterisk bounty: SIP simultaneous
Paul Mahler
pmahler at signate.com
Sun Jul 11 08:42:26 MST 2004
You are confused about what a SIP session is and what a SIP session does.
SIP, session initiation protocol, controls an RTP, real time protocol,
session between two IP endpionts. The end points have to have unique IP
addresses for the session to run. The unique SIP registration is how * finds
a UNIQUE endpoint.
You don't want SIP to solve your problem, you want * to solve your problem.
You are asking for this SIP "feature" because you are confused as to how SIP
and * work, and how they work together.
You can easily fix your business problem with *, but not with mechanism you
are asking for. You should spend your money on getting a copy of each of the
two books that are now available and learn *. Then it will be clear to you
that you don't really want what you are asking for.
Paul
Paul Mahler
pmahler at signate.com
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> Kannaiyan Natesan
> Sent: Sunday, July 11, 2004 1:15 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
>
> 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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