asterisk users - Aug 2007 - OT - P-asserted-identity and remote id

Hi,

The case I'm working on is :
- a call comes from PSTN to a given extension (say 122)
- a user picks the call up (dialing *8122)  from another extension (say 240)
using a SIP hardphone
- the hardphone (he one with 240 extension) displays the dialed string (here
*8122) instead of original caller-id.

This is logical but I would like to change this default behaviour so that
original caller-id is displayed along or instead of dialed string.

SIP hardphone vendor says it could be either done with :
1. SIP MESSAGE
2. SIP P-asserted-identity
(Unfortunately both way are not supported yet by this phone vendor but
that's another story).

Anyway, I tried to understand what SIP P-asserted-identity is and how it
relates to my case.
I can't see any relation.

Can anyone explain ?

My thoughts are :
- when replying 200 OK to SIP INVITE (from 240 extension), Asterisk server
has to add a field P-Asserted-Identity in SIP header filled with original
caller-id (using SipAddHeader ?)
- when receiving such SIP message, P-Asserted-Identity-enabled SIP phone
would display P-Asserted-Identity field data instead of what it uses to
display
- is this correct ?
- and what if phone also supports caller-name display (phone has embedded
directory and when caller-id matches with a directory entry, it replaces
caller-id with caller-name) ? Shall I understand it will use
P-Asserted-Identity data for directory lookup instead of other SIP header
field ?


Regards
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You can add the header the vendor is suggesting in asterisk as follows;

 

exten => #####,1,SipAddHeader(P-Asserted-Identity:
<sip:${CALLERID(num)}@hostname>)

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, August 08, 2007 2:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT - P-asserted-identity and remote id

 

Hi,

The case I'm working on is :
- a call comes from PSTN to a given extension (say 122)
- a user picks the call up (dialing *8122)  from another extension (say
240) using a SIP hardphone
- the hardphone (he one with 240 extension) displays the dialed string
(here *8122) instead of original caller-id. 

This is logical but I would like to change this default behaviour so
that original caller-id is displayed along or instead of dialed string.

SIP hardphone vendor says it could be either done with :
1. SIP MESSAGE 
2. SIP P-asserted-identity
(Unfortunately both way are not supported yet by this phone vendor but
that's another story).

Anyway, I tried to understand what SIP P-asserted-identity is and how it
relates to my case. 
I can't see any relation.

Can anyone explain ?

My thoughts are :
- when replying 200 OK to SIP INVITE (from 240 extension), Asterisk
server has to add a field P-Asserted-Identity in SIP header filled with
original caller-id (using SipAddHeader ?) 
- when receiving such SIP message, P-Asserted-Identity-enabled SIP phone
would display P-Asserted-Identity field data instead of what it uses to
display
- is this correct ?
- and what if phone also supports caller-name display (phone has
embedded directory and when caller-id matches with a directory entry, it
replaces caller-id with caller-name) ? Shall I understand it will use
P-Asserted-Identity data for directory lookup instead of other SIP
header field ? 


Regards




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