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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>You can add the header the vendor is
suggesting in asterisk as follows;</span></font></p>
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10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>exten => #####,1,SipAddHeader(P-Asserted-Identity:
<sip:${CALLERID(num)}@hostname>)</span></font></p>
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10.0pt;font-family:Arial;color:navy'> </span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Olivier<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 08, 2007
2:51 AM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] OT -
P-asserted-identity and remote id</span></font></p>
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12.0pt'> </span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Hi,<br>
<br>
The case I'm working on is :<br>
- a call comes from PSTN to a given extension (say 122)<br>
- a user picks the call up (dialing *8122) from another extension (say
240) using a SIP hardphone<br>
- the hardphone (he one with 240 extension) displays the dialed string (here
*8122) instead of original caller-id. <br>
<br>
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.<br>
<br>
SIP hardphone vendor says it could be either done with :<br>
1. SIP MESSAGE <br>
2. SIP P-asserted-identity<br>
(Unfortunately both way are not supported yet by this phone vendor but that's
another story).<br>
<br>
Anyway, I tried to understand what SIP P-asserted-identity is and how it
relates to my case. <br>
I can't see any relation.<br>
<br>
Can anyone explain ?<br>
<br>
My thoughts are :<br>
- 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 ?) <br>
- 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<br>
- is this correct ?<br>
- 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
? <br>
<br>
<br>
Regards<br>
<br>
<br>
</span></font></p>
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