Re: Update the LCD with the callee's name after dialing

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The client needs to support the Remote-Party-ID SIP header. If you

want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.

Ryan

--
Ok, i did a sniffing and the header is added by asterisk, but marked as unrecognised header.
So this means the client is not able to deal with it, correct?

pHSIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.149:1245;branch=z9hG4bK-d8754z-1f09b71f6073d335-1---d8754z-;received=192.168.1.149;rport=1245
From: <sip:1000@xxxxxxxxxxxx:5060>;tag=485db579
To: <sip:2000@xxxxxxxxxxxx:5060>;tag=as71ddd62f
Call-ID: NGJjNDEyNmQxNGUyYTI1YTdhN2MwZWQwYzVhMTA2ZjA.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:2000@xxxxxxxxxxxx>
Date: Sat, 03 Jul 2010 09:25:11 GMT
Remote-Party-ID: "Test2" <sip:2000@xxxxxxxxxxxx:5060>;party=called;id-type=subscriber;screen=yes
Expert Info (Note/Undecoded): Unrecognised SIP header (Remote-Party-ID)
Message: Unrecognised SIP header (Remote-Party-ID)
Severity level: Note
Group: Undecoded
-- 
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