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

[Ryan Wagoner]

On Tue, Jul 6, 2010 at 10:19 AM,  <unserossi at 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:2000 at 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 at 192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP>
> To: "Callee Name" <sip:2000 at 192.168.1.10:5060>
> From: "Caller Name" <sip:1000 at 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