[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

> 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



-- 

But if the Remote-Party-ID is set or not can only be checked by sniffing with Wireshark or another sniffer.
It can not be checked by using "sip set debug on" in Asterisk. Correct?
Because there I cannot see anything added.

 
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