Diversion header missing

Distro Discussion & Help
1

freebpx disro 14.0.1.20 asterisk 14.6.0
trying to get inbound caller id to display when call is forwared back outside system
carrier requires from:
enabled diversion header (settings-advanced) but sip invite still has no diversion header
tried p-asseted-id - but the header had the inbound did as the number -
what am i missing ??

call comes into 5612223333 - forwards on ext 1234 to 9545551234

call comes into 5612223333 - forwards on ext 1234 to 9545551234

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.200.52:5060;branch=z9hG4bK36e8a24d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as6e555b93
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: FPBX-14.0.1.20(14.6.0)
Date: Mon, 30 Oct 2017 19:55:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “5612223333” sip:[email protected]
Content-Type: application/sdp
Content-Length: 350

v=0
o=root 1984729299 1984729300 IN IP4 10.1.200.52
s=Asterisk PBX 14.6.0
c=IN IP4 10.1.200.52
t=0 0
m=audio 15080 RTP/AVP 0 9 8 107 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

2

A coherent explanation.

Just to take a stab at it, check your caller ID options on your trunk. Are you forcing the trunk CID and/or blocking foreign CID?

1 Like
3

had no caller id on trunk
was allowing any callerr id
tried variations -but divert header is not inserting -
carrier requires from field to be a valid did in the sip trunk group
which i have to argue with them a little
if the diversion header is there - they should honor that then the from field would be the forwarded caller id number
I can get that portion working - no problem -
just cant seem to get the diversion header inserted

4

Hello @c3cloud,

Did you enabled the diversion header generation in the advanced settings module?

image.png1753×37 2.97 KB

Thank you,

Daniel Friedman
Trixton LTD.

5

definitely enabled - generate diversion headers - yes

6

Hello @c3cloud,

If it is enabled, you need to use it with followme module. If you will not do that you will have to generate your customized diversion header through the macro-dialout-trunk-predial-hook context.

You can use a virtual extension with a follome enabled to generate a diversion header as well.

Thank you,

Daniel Friedman
Trixton LTD.

7

tried it with the follow me feature - still no luck
will have to mess with the macro - not my best forte…
thanks for pointing me in that direction though

8

Hello @c3cloud,

Can you share the sip trace with the followme enabled?

Thank you,

Daniel Friedman
Trixton LTD.

9

without copying a ton of sip headers and editing them
the outbound call with find me followme is as follows
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.200.52:5060;branch=z9hG4bK623a8e29;rport
Max-Forwards: 70
From: sip:[email protected] ;tag=as7c43a361
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.1.20(14.7.0)
Date: Mon, 06 Nov 2017 21:52:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “caller id name” sip:[email protected];party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 348

    v=0
    o=root 734574809 734574809 IN IP4 10.1.200.52
    s=Asterisk PBX 14.7.0
    c=IN IP4 10.1.200.52
    t=0 0
    m=audio 13046 RTP/AVP 9 0 8 107 3 101
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:107 opus/48000/2
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:20
    a=sendrecv
10

You need to show the dialplan execution from the console. This is a SIP header that needs to be added, the adding of that header will be in the dialplan process. Need to see if the header is actually being set or the diversion setting is even being checked or handled.

11

Hello @c3cloud,

There is no generation of a diversion header, only a RPID is being sent.

Set your extensions and trunks to send PAI headers (sendrpid=pai) and check your advanced settings again.

Thank you,

Daniel Friedman
Trixton LTD.

12

Hi, did you solved it? I have exactlly the same problem. It used to work ok for me but suddenly stopped.