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:1...@192.168.1.10:5060>;tag=485db579 To: <sip:2...@192.168.1.10:5060>;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:2...@192.168.1.10> Date: Sat, 03 Jul 2010 09:25:11 GMT Remote-Party-ID: "Test2" <sip:2...@192.168.1.10:5060>;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
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
