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<div style="BACKGROUND-COLOR: #fff; MARGIN: 0px; FONT-FAMILY: Tahoma, Verdana, Arial, Sans-Serif; COLOR: #000; FONT-SIZE: 12px" id=AOLMsgPart_0_69a6c82a-0897-41e7-9ef6-6410336ed254><PRE style="FONT-SIZE: 9pt"><TT>>>> The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
>>> compile but need to be tested to verify that they work. I have the
>>> 1.6.2.9 version in production and plan to put the 1.6.1.20 version in
>>> sometime this weekend.
>>>
>>> In you are just using Asterisk in the dialplan you can set the called
>>> remote party id with something like below. Otherwise check out the
>>> previous FreePBX 2.7 patch.
>>>
>>> exten =>
>>>
>>>
>>> 100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
>>>
>>> Ryan
>>
>> If you installed Asterisk from source you just need to patch and
>> recompile / install.
>>
>> cd asterisk-version
>> patch -p1 < ../asterisk-verson-called-
>> rpid.patch
>> make install
>>
>> Otherwise if your using trixbox, etc you would probably want to grab
>> their SRPMS, add the patch to the spec file, and rebuild them. However
>> that is outside of the scope of this mailing list.
>>
>> Ryan
>
> Which version of Asterisk? The patches were made against the latest
> releases. If you are running an earlier version you might need to
> manually patch your install.
>
> Ryan
>
> --
>
> Version 1.6.1.20
>
> But it was my individual problem. Installing from scratch solved the
> patching issue.
>
> Now the application SIPCalledRPID is active and gets executed but i still
> don't get the name of the called person
>
> on my display. Maybe this is client dependent? I am using 3CX Softphone. Or
> is somethins else missing?
>
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
--
<FONT size=2>Just to make sure that we are talking about the same issue.</FONT></TT></PRE><PRE style="FONT-SIZE: 9pt"><TT><FONT size=2>What I want is that when two users are registered at the same peer that </FONT></TT></PRE><PRE style="FONT-SIZE: 9pt"><TT><FONT size=2>when user A calls user B user A gets the name of user B displayed on his client.<PRE style="FONT-SIZE: 9pt"><TT><FONT size=2>Is this what you are trying to fix with the patch? </FONT></TT></PRE></FONT></TT></PRE><PRE style="FONT-SIZE: 9pt"><TT><FONT size=2>Because from my understanding as an absolute newbie to SIP and Asterisk, the header </FONT></TT></PRE><PRE style="FONT-SIZE: 9pt"><TT><FONT size=2>should already contain the let's call it "displayname" and look something like</FONT></TT></PRE><PRE style="FONT-SIZE: 9pt"><TT><FONT size=2>INVITE sip:2000@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP>
To: "Callee Name" <sip:2000@192.168.1.10:5060>
From: "Caller Name" <sip:1000@192.168.1.10:5060>;tag=cf41cd30
</FONT></TT></PRE><PRE style="FONT-SIZE: 9pt"><TT><FONT size=2 face=Tahoma>according to SIP rfc 3261 <A href="http://tools.ietf.org/html/rfc3261">http://tools.ietf.org/html/rfc3261</A></FONT></TT></PRE></div>
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