[Asterisk-Users] New Asterisk bounty: SIP simultaneous

Sun Jul 11 10:57:45 MST 2004

As Daniel Says, Bounty stands.

I cannot explain to you anymore. I'm sorry.
Please read more what SIP can do with SER.

-Kannaiyan.

----- Original Message -----
From: "Paul Mahler" <pmahler at signate.com>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, July 11, 2004 4:42 PM
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

> 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/
> > > >
> > > > _______________________________________________
> > > > 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
> > >
> > > _______________________________________________
> > > 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
> >
>
> _______________________________________________
> 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
>
>