On Tue, Jul 6, 2010 at 10:19 AM, <unsero...@aol.com> wrote: >>>> 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 > > -- > Just 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 > Via: SIP/2.0/TCP > 192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport > Max-Forwards: 70 > Contact: > <sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP> > To: "Callee Name" <sip:2...@192.168.1.10:5060> > From: "Caller Name" <sip:1...@192.168.1.10:5060>;tag=cf41cd30 > > according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261 >
Yes that is what the patch addresses. The phones will only display the name of the called extension if Remote-Party-ID or P-Asserted-Identity is set. Ryan -- _____________________________________________________________________ -- 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