[Asterisk-Users] Multiple Line Appearances

[John Todd]

My point in my last paragraph was that you don't want to do this on 
the user-agent side; you want to control this on the server side.

To use asterisk parlance:

exten => 1234,1,Dial(SIP/jane&SIP/bill)

This means that when extension 1234 is called, that the (single) 
phones named "jane" and "bill" will ring.  Those SIP peers are 
defined in sip.conf.

While the SIP specification allows one to have multiple endpoints 
registering with the same authentication name and password (and thus, 
the same "identity") this is IMHO not a good design, since there is 
no accountability for where calls actually went or came from.  I 
suppose if you are a completely "open" network then this is OK, but 
anywhere that there are monetary expenses associated with calls, this 
will quickly lead to heartache and woe.  In any case, Asterisk only 
recognizes the most recent registration that it has seen for a 
particular identity, so every few seconds you'd get a tug-of-war 
going on with multiple identities registering to the same sip peer 
entry.

SER can do both methods (multiple phones mapped to 1 identity, or 
multiple phones mapped to multiple identities) but this is the 
Asterisk mailing list, not the SER mailing list.  :-)


JT


At 12:00 PM -0600 1/31/04, John Baker wrote:
>How were you able to integrate this with asterisk?  Or did you drop
>asterisk in favor of ser?
>
>John
>
>On Thu, 2004-01-29 at 12:44, John Todd wrote:
>>  At 12:20 PM -0500 1/29/04, Jeremy McNamara wrote:
>>  >MLS Drop for SysAdmin wrote:
>>  >
>>  >>Has anyone successfully implemented concurrent appearance of the
>>  >>same PBX extension on multiple SIP phones?
>>  >>
>>  >>When using Cisco 7960s under call manager, you can have several
>>  >>phones with the same line appearance, but the first user to seize a
>>  >>line makes it inaccessible to other phones.
>>  >>
>>  >>Under SIP operation it seems as though this is not possible, but we
>>  >>don't see group ringing definable for SIP extensions.
>>  >
>>  >
>>  >It is my understanding that Cisco didn't bother implementing this
>>  >functionality into their SIP firmware. However, as you have
>>  >described, this feature does work when using CCM.
>>  >
>>  >chan_skinny (and chan_sccp - which btw, will become the same channel
>>  >driver soon) will eventually support this feature.
>>  >
>>  >If you (anyone?) have any motivation for Theo and myself to make
>>  >Asterisk's SCCP support go to the top of our to-do lists, please
>>  >contact either one of us off-list.
>>  >
>>  >Jeremy McNamara
>>
>>  The Cisco phones with SIP support this just fine; it's not a problem
>>  for the endpoints, it's a problem for the SIP registrar.
>>
>>  The phones will happily send out the same authentication
>>  name/password pair all day long to the server. The server must be
>>  smart enough to then map those multiple registrations to a single
>>  "number." Asterisk does not support this feature at this time.
>>
>>  If you want to use this trick, try SER, as I have had multiple
>>  devices with the same registration data register against SER.  When
>>  an INVITE is passed into the system, all the phones automatically
>>  ring and the first pickup gets the call.
>>
>>  Typically, you'd want to do this type of multi-number mapping back on
>>  the server, anyway - it's not a good idea to have multiple endpoints
>>  registering with the same auth data, but you can do it if you really,
>>  really want - just not with Asterisk.
>>
>  > JT
>>