Also I just made a few tests. And it appears it works, under certain conditions…
-- Executing [<DialedNumber>@<privateInfoRemoved>-outbound-SIPDevices:2] Set("PJSIP/<privateInfoRemoved>-802-00000000", "CONNECTEDLINE(name)=Hello World") in new stack
-- Executing [<DialedNumber>@<privateInfoRemoved>-outbound-SIPDevices:3] Set("PJSIP/<privateInfoRemoved>-802-00000000", "CONNECTEDLINE(num)=22222222") in new stack
<--- Transmitting SIP response (696 bytes) to UDP:<sipPhoneIP>:40561 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP <sipPhoneIP>:40561;rport=40561;received=<sipPhoneIP>;branch=z9hG4bK-mzw9scxl1yi7
Call-ID: 313630373431343930343230333231-1xmymoyfzuus
From: "Test (802)" <sip:<privateInfoRemoved>-802@<sipServer>>;tag=nn5kakvfsd
To: <sip:70704040@<sipServer>;user=phone>;tag=20216562-a90a-4b18-96d3-2b416970410e
CSeq: 1 INVITE
Server: Asterisk
Contact: <sip:<sipServerIP>:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER
Remote-Party-ID: "Hello World" <sip:22222222@<sipServer>;user=phone>;party=called;privacy=off;screen=no
Content-Length: 0
I did however have to add the following to the device template in pjsip_wizard.conf
endpoint/send_connected_line = yes
endpoint/connected_line_method = invite
endpoint/send_rpid = yes
endpoint/rpid_immediate = yes
I added all of them at once, so don’t know if they are all needed. But at least ONE of them is, as the remote party header did not appear with just the set lines in the dial plan.
The SNOM 720, I used to test the call, however, just ignored the information. 