[asterisk-users] Update the LCD with the callee's name after dialing

[unserossi at aol.com]

>>> 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
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:2000 at 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:1000 at 192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP>
o: "Callee Name" <sip:2000 at 192.168.1.10:5060>
rom: "Caller Name" <sip:1000 at 192.168.1.10:5060>;tag=cf41cd30
according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100706/4a30a971/attachment.htm