[asterisk-dev] pjsip: 180 Ringing contains wrong info in Remote-Party-ID

[Steve Murphy]

Hello--

I've got complaints that the phones are presenting the wrong info when
making an outgoing call... instead of displaying the called party info,
it's displaying the caller's info, which is highly uninteresting. I've been
looking at the behavior with Yealink phones, but I'm told that ALL phones
have the problem, and comparing with the sip channel driver.

I'm working with asterisk (and pjsip) at version 13.15.0, so this is pretty
much current behavior.

I traced it down to the 180 Ringing message sent to the phone from
Asterisk, in the course of making an outgoing call from the Yealink, in
this case, to another extension on the same phone system.

In the old chan_sip world, I see this:

[May 13 13:10:58] <--- Transmitting (NAT) to 67.215.23.186:28762 --->
[May 13 13:10:58] SIP/2.0 180 Ringing
[May 13 13:10:58] Via: SIP/2.0/UDP 192.168.134.126:5060
;branch=z9hG4bK1785363097;received=67.291.23.186;rport=28762
[May 13 13:10:58] From: "Steve Murphy" <sip:nvl19049 at 190.190.190.190:5060
>;tag=2559859725
[May 13 13:10:58] To: <sip:767 at 190.190.190.190:5060>;tag=as0a66b2c7
[May 13 13:10:58] Call-ID: 0_762068959 at 192.168.134.126
[May 13 13:10:58] CSeq: 2 INVITE
[May 13 13:10:58] Server: nexVortex Inc Hosted 3.0 PBX
[May 13 13:10:58] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 13 13:10:58] Supported: replaces, timer
[May 13 13:10:58] Contact: <sip:767 at 190.190.190.190:5060>
[May 13 13:10:58] Remote-Party-ID: "Shifting Sands" <sip:767 at 190.190.190.190
>;party=called;privacy=off;screen=no
[May 13 13:10:58] Content-Length: 0
[May 13 13:10:58]

Note, that Asterisk serves up callerid info from the target extension in
this header, providing not only the number of the target extension, but the
callerid NAME info, also, which is pretty nice!

​But, in the PJSIP world, I see this instead (on a different test system):

[May 13 08:21:59] <--- Transmitting SIP response (597 bytes) to UDP:
192.168.134.102:5060 --->
[May 13 08:21:59] SIP/2.0 180 Ringing
[May 13 08:21:59] Via: SIP/2.0/UDP 192.168.134.102:5060
;rport=5060;received=192.168.134.102;branch=z9hG4bK1705376406
[May 13 08:21:59] Call-ID: 0_1685072057 at 192.168.134.102
[May 13 08:21:59] From: "Steve" <sip:t12 at 192.168.134.227>;tag=3119644064
[May 13 08:21:59] To: <sip:102 at 192.168.134.227
>;tag=c7988cae-0380-49b4-84e6-0a03b656ab85
[May 13 08:21:59] CSeq: 2 INVITE
[May 13 08:21:59] Server: nexVortex SoupedUp Asterisk Hybrid
[May 13 08:21:59] Contact: <sip:192.168.134.227:57969>
[May 13 08:21:59] Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK,
BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
[May 13 08:21:59] Remote-Party-ID: "Steve" <sip:101 at 192.168.134.227
>;privacy=off;screen=no
[May 13 08:21:59] Content-Length:  0


​In this instance, it just looks like the rpid is a copy of the "From:"
header. This isn't so interesting, as I already know my own name and
extension number!

I traced this down to the add_rpid_header() func in the
res/res_pjsip_caller_id module... but I suspect that the connected line
updates play a role here, and I'm too much a nube to know where the "right"
information is.

​Am I hallucinating? Got a bad config? Or is there a bug here?

murf
​

-- 

Steve Murphy
ParseTree Corporation
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