It's not what SIP does with SER, it's what SER does with SIP.
Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Kannaiyan Natesan > Sent: Sunday, July 11, 2004 9:58 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous > > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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 > > [EMAIL PROTECTED] > > Signate, LLC > > 665 Third Street > > Suite 100 > > San Francisco, CA > > 94107-1901 > > > > Asterisk Services and Training > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of > > > Kannaiyan Natesan > > > Sent: Sunday, July 11, 2004 1:15 AM > > > To: [EMAIL PROTECTED] > > > 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" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > 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: [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] Behalf Of > > > > Kannaiyan > > > Natesan > > > > Sent: 11 July 2004 09:51 > > > > To: [EMAIL PROTECTED] > > > > 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" <[EMAIL PROTECTED]> > > > > To: <[EMAIL PROTECTED]> > > > > 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 > > > > > [EMAIL PROTECTED] > > > > > 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 > > > > [EMAIL PROTECTED] > > > > 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 > > > > [EMAIL PROTECTED] > > > > 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 > > > [EMAIL PROTECTED] > > > 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 > > [EMAIL PROTECTED] > > 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 > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
