Pjsip diversion

@mitja wrote:

Hello

I was testing a FreePbx 13 with pjSip and I have some strange behavior.
On call forwarding, there is no diversion header, I have enabled diversion in advanced settings and i can see in dial plan that context [sub-diversion-header] is executed (PJSIP_HEADER(add,Diversion)=;privacy=off;screen=no;reason=unconditional)), but in sip trace is missing.

INVITE sip:4888888@192.168.148.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.148.9:5060;rport;branch=z9hG4bKPj502tghnXQSuXRPhSv3RLN0yeHhv4vOmo
From: sip:4999999@192.168.148.9;tag=BEUrTEUcYKh3whPGHhEx5IYIOuYjdJ.1
To: sip:4888888@192.168.148.10
Contact: sip:asterisk@192.168.148.9:5060
Call-ID: Utl9Q1tX0PtiMkMQkC8Mk-1F.JQDe7gp
CSeq: 29621 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: sip:4999999@192.168.148.9
Remote-Party-ID: sip:4999999@192.168.148.9;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 13.12.1
Content-Type: application/sdp
Content-Length: 263

v=0
o=- 1008816852 1008816852 IN IP4 192.168.148.9
s=Asterisk
c=IN IP4 192.168.148.9
t=0 0
m=audio 15088 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Until now I was using chan_sip and it was working with no problems

BR
Mitja

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