>>> 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.
>
> RyanWhich 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 ant to verify the header is being added run tcpdump and analyze it ith Wireshark. I know that Polycom phones have support for this. I ust put a modified version of the Asterisk 1.6.1 patch into roduction for 25 Polycom phones, soon to be 150 phones. I changed the eturn -1 to return 0 so that the call continues even if they IPCalledRPID args are invalid. Ryan -- ust to make sure that we are talking about the same issue. What I want is that when two users are registered at the same peer that when user A calls user B user A gets the name of user B displayed on his client. Is this what you are trying to fix with the patch? Because from my understanding as an absolute newbie to SIP and Asterisk, the header should already contain the let's call it "displayname" and look something like INVITE sip:2...@192.168.1.10:5060 SIP/2.0 ia: SIP/2.0/TCP 192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport ax-Forwards: 70 ontact: <sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP> o: "Callee Name" <sip:2...@192.168.1.10:5060> rom: "Caller Name" <sip:1...@192.168.1.10:5060>;tag=cf41cd30 according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261
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