Hello,
I have problem with blind transfer on my PBXAct. I know from provider that I need to generate diversion header in SIP but it seems that it’s not generated.
In advanced setting “Generate Diversion Headers” is on:
For extension and SIP trunk:
sendrpid=pai
trustrpid=yes
But in SIP debug to provider I can’t see field divension (there is an PAI but no divension).
Part of my SIP debug during blind transfer is as follow:
Audio is at 14208
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to YY.YY.YY.YY:5283:
INVITE sip:[email protected]:5283;user=phone SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK42b1e686
Max-Forwards: 70
From: “AAAAAAAAA” sip:[email protected];tag=as008e0059
To: sip:[email protected]:5283;user=phone
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: PBXact-13.0.194.2
Date: Thu, 07 May 2020 11:30:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “AAAAAAAAA” sip:[email protected]
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 2125675531 2125675531 IN IP4 YY.YY.YY.YY
s=Asterisk PBX
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 14208 RTP/AVP 8 9 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
Where is a problem?
What should I change to generate that field in my SIP?
Current Asterisk Version: 13.29.2
Best regards,
Andcza