VitalPBX not sending Diversion header on call forward

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?

Hello @Equi-tel and welcome to the community!

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

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

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.

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

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