<font color='black' size='2' face='arial'>>>>> The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both<br>
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>>>> compile but need to be tested to verify that they work. I have the<br>
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>>>> 1.6.2.9 version in production and plan to put the 1.6.1.20 version in<br>
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>>>> sometime this weekend.<br>
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>>>> In you are just using Asterisk in the dialplan you can set the called<br>
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>>>> remote party id with something like below. Otherwise check out the<br>
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>>>> previous FreePBX 2.7 patch.<br>
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>>>> 100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)<br>
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>>>> Ryan<br>
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>>> If you installed Asterisk from source you just need to patch and<br>
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>>> recompile / install.<br>
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>>> cd asterisk-version<br>
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>>> patch -p1 < ../asterisk-verson-called-<br>
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>>> Otherwise if your using trixbox, etc you would probably want to grab<br>
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>>> their SRPMS, add the patch to the spec file, and rebuild them. However<br>
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>>> that is outside of the scope of this mailing list.<br>
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>>> Ryan<br>
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>> Which version of Asterisk? The patches were made against the latest<br>
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>> releases. If you are running an earlier version you might need to<br>
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>> manually patch your install.<br>
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>> Ryan<br>
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>> --<br>
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>> Version 1.6.1.20<br>
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>> But it was my individual problem. Installing from scratch solved the<br>
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>> patching issue.<br>
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>> Now the application SIPCalledRPID is active and gets executed but i still<br>
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>> don't get the name of the called person<br>
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>> on my display. Maybe this is client dependent? I am using 3CX Softphone.<br>
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>> Or<br>
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>> is somethins else missing?<br>
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> The client needs to support the Remote-Party-ID SIP header. If you<br>
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> want to verify the header is being added run tcpdump and analyze it<br>
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> with Wireshark. I know that Polycom phones have support for this. I<br>
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> just put a modified version of the Asterisk 1.6.1 patch into<br>
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> production for 25 Polycom phones, soon to be 150 phones. I changed the<br>
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> return -1 to return 0 so that the call continues even if they<br>
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> SIPCalledRPID args are invalid.<br>
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> Ryan<br>
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> Just to make sure that we are talking about the same issue.<br>
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> What I want is that when two users are registered at the same peer that<br>
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> when user A calls user B user A gets the name of user B displayed on his<br>
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> client.<br>
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> Is this what you are trying to fix with the patch?<br>
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> Because from my understanding as an absolute newbie to SIP and Asterisk, the<br>
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> header<br>
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> should already contain the let's call it "displayname" and look something<br>
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> like<br>
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> INVITE sip:<a href="mailto:2000@192.168.1.10">2000@192.168.1.10</a>:5060 SIP/2.0<br>
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> Via: SIP/2.0/TCP<br>
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> 192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport<br>
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> Max-Forwards: 70<br>
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> Contact:<br>
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> <sip:<a href="mailto:1000@192.168.1.149">1000@192.168.1.149</a>:3823;rinstance=8f3067c0aac0abc4;transport=TCP><br>
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> To: "Callee Name" <sip:<a href="mailto:2000@192.168.1.10">2000@192.168.1.10</a>:5060><br>
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> From: "Caller Name" <sip:<a href="mailto:1000@192.168.1.10">1000@192.168.1.10</a>:5060>;tag=cf41cd30<br>
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> according to SIP rfc 3261 <a target="_blank" href="http://tools.ietf.org/html/rfc3261">http://tools.ietf.org/html/rfc3261</a><br>
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Yes that is what the patch addresses. The phones will only display the<br>
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name of the called extension if Remote-Party-ID or P-Asserted-Identity<br>
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Ryan<br>
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But if the Remote-Party-ID is set or not can only be checked by sniffing with Wireshark or another sniffer.<br>
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It can not be checked by using "sip set debug on" in Asterisk. Correct?<br>
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Because there I cannot see anything added.<br>
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<pre style="font-size: 9pt;"><tt>-- <br>
I am sorry, my fault. It is added and I can see it in Asterisk sip debug.<br>
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But comparing the Remote-Party-ID Header of a (displayed) caller and a (not displayed) callee looks a bit different.<br>
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Remote-Party-ID: "Callee" <sip:2000@192.168.1.10:5060>;party=called;id-type=subscriber;screen=yes<br>
Remote-Party-ID: "Caller" <sip:1000@192.168.1.10>;privacy=off;screen=yes<br>
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Could maybe this be the reason why it does not work for me?<br>
Sorry if I ask stupid questions but this feature is quite important for me.<br>
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