[SIPForum-discussion] Calling Party Name

[Garron, James]

Yudia,

>From RFC324 section 8.4:

RFC 2543 [10] specified that placing a user on hold was accomplished by
setting the connection address to 0.0.0.0. Its usage for putting a call
on hold is no longer recommended, since it doesn't allow for RTCP to be
used with held streams, doesn't work with IPv6, and breaks with
connection oriented media. However, it can be useful in an initial offer
when the offerer knows it wants to use a particular set of media streams
and formats, but doesn't know the addresses and ports at the time of the
offer. Of course, when used, the port number MUST NOT be zero, which
would specify that the stream has been disabled. An agent MUST be
capable of receiving SDP with a connection address of 0.0.0.0, in which
case it means that neither RTP nor RTCP should be sent to the peer.

-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of yudia
Sent: Tuesday, April 14, 2009 8:09 PM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Calling Party Name

Hi all,

i still having having trouble in displaying the calling party name,
because we only receive calling party number between sip users.
We are using PAI header, and even tough we did found the user name in
trace log, but still cannot display CP-Name.

Thank you for any inputs.

Yudia


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