VitalPBX not sending Diversion header on call forward

VitalPBX
1

Hi,
I am trying to forward an external Caller to an external Callee. When I check the trace in sngrep, I do not see a diversion header from my VitalPBX instance and my SIP provider drops the call because I send a number that is not in my sip account. I have the option “Send Diversion Header” checked in “Device Profiles”. In my extension, the “CallerID On Diversions” is set to CallerID. I also tried using the “Follow me” option with the same output.
Is there a way to send a diversion header only when there’s a call forward or follow me?

2

Hello @telsupport and welcome to the community!

Can you please share screenshot of your Trunk configuration as well as a full call trace via pastebin.com

3

Hi PitzKey,
I’m not quite sure on how to use Pastebin so heres in plain text the trunk’s configuration :

[T15_XXXXXXXXXX](p1)
type=endpoint
dtmf_mode=rfc4733
transport=transport-udp-3baa003c2d301de89c68
context=trk-16-in
allow=!all,ulaw
language=fr_FR
aors=T15_XXXXXXXXXX
outbound_auth=T15_XXXXXXXXXX-oauth
from_domain=exemple.ca
trust_id_inbound=no
trust_id_outbound=no
t38_udptl=no
t38_udptl_ec=none
t38_udptl_maxdatagram=0
fax_detect=no
fax_detect_timeout=0
t38_udptl_nat=no
t38_udptl_ipv6=no
[T15_XXXXXXXXXX-oauth]
type=auth
auth_type=userpass
username=XXXXXXXXXX
password=&&&&&&&&&&&&&&&

[T15_XXXXXXXXXX](p1-aor)
type=aor
max_contacts=2
contact=sip:exemple.ca
qualify_frequency=30
qualify_timeout=3

[T15_XXXXXXXXXX-reg-1]
type=registration
endpoint=T15_XXXXXXXXXX
transport=transport-udp-3baa003c2d301de89c68
outbound_auth=T15_XXXXXXXXXX-oauth
server_uri=sip:exemple.ca
client_uri=sip:XXXXXXXXXX@exemple.ca
retry_interval=60
max_retries=10
forbidden_retry_interval=10
expiration=3600
line=yes
auth_rejection_permanent=no

And here’s a capture from sngrep, it is the invite from VitalPBX to our provider :

INVITE sip:[ExternalCalleeID]@equi-tel.ca SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPja1e248fa-bb1e-462f-bf47-29368f51760e
From: <sip:[ExternalCallerID]@exemple.ca>;tag=daf9c468-e769-4a29-b3d5-f625e1aa2aac
To: <sip:[ExternalCalleeID]@exemple.ca>
Contact: <sip:asterisk@XXX.XXX.XXX.XXX:5060>
Call-ID: 3978dcb4-1df2-41cd-92f6-54c09079827e
CSeq: 3927 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: VitalPBX
Content-Type: application/sdp
Content-Length:   243

v=0
o=- 1917253720 1917253720 IN IP4 205.233.212.190
s=Asterisk
c=IN IP4 205.233.212.190
t=0 0
m=audio 13406 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
4

Please share a screenshot of your Trunk configuration.

And a call trace (asterisk logs) via pastebin.com. Asterisk logs is not easy to read on the forums, using pastebin helps us opening raw and pointing out line numbers.

5

Hi,

Here’s the Asterisk logs of a test I did this morning ; Logs 2023-06-15 - Pastebin.com

Here’s the Trunk configs ; Trunk config - Pastebin.com

6

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