SIP Trunk "From" and "Route" fields are incorrect

I am connecting my Asterisk PBX to cloud-PBX provider 8x8 via a SIP tie trunk.

Calling from Asterisk to 8x8 does not work. This is the issue that I would like to solve. The call is placed and immediately ends. 8x8 sends back a SIP 482 message, indicating a Loop has been detected.

Calling from 8x8 to Asterisk does work, except for one-way-audio.

I have been tinkering with whatever I can think of and I am now at a dead end. Please help!

Note - 8x8 engineering has identified that the Route field is causing this problem. How to fix this is my question…

Asterisk Extension = 10393
8x8 Extension = 10001

SIP Trunk requirements from 8x8

RURI needs to be in the following format:[extension_number]@[inbyochealthfirstsharedpserv.8x8.com]
(This is now correct, thanks to input received on a previous thread.)

8x8 Proxy Server for SIP trunk: ssbctrunk-us.8x8.com:5500

pjsip.conf
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.0.0.0/8
local_net=172.16.0.0/12
local_net=192.168.0.0/16
external_media_address=44.216.61.119
external_signaling_address=44.216.61.119

[8x8]
type=endpoint
transport=transport-udp-nat
context=from-8x8
outbound_proxy=sip:ssbctrunk-us.8x8.com:5500;lr
disallow=all
allow=ulaw
identify_by=ip
aors=8x8

[8x8]
type=identify
endpoint=8x8
match=129.151.80.0/25
match=158.101.41.0/25

[8x8]
type=aor
contact=sip:inbyochealthfirstsharedpserv.8x8.com
qualify_frequency=30

[10393]
type=endpoint
context=softphone
transport=transport-udp-nat
disallow=all
allow=ulaw
auth=auth10393
aors=10393
force_rport=yes
rtp_symmetric=yes
direct_media=yes

[auth10393]
type=auth
auth_type=userpass
password=xxxxxxxx
username=10393

[10393]
type=aor
max_contacts=1
remove_existing=yes

PCAP of Invite from failed call

The “From” field shows my Asterisk server private IP, not the public IP.
The “Route” field should show my public IP address, not the 8x8 Proxy Server.

Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:10001@inbyochealthfirstsharedpserv.8x8.com SIP/2.0
Method: INVITE
Request-URI: sip:10001@inbyochealthfirstsharedpserv.8x8.com
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 44.216.61.119:5060;rport;branch=z9hG4bKPjSBPb9hQaZQAbPKKqZA.It-aEan2F.-Cb
From: sip:10393@10.254.40.69;tag=dY-grsGPLNOeP9fQenEFc-yz3k6U-DIz
To: sip:10001@inbyochealthfirstsharedpserv.8x8.com
Contact: sip:asterisk@44.216.61.119:5060
Call-ID: lzyJFyAvr8IE5gwuIqoqGNIAaBiNCdcA
[Generated Call-ID: lzyJFyAvr8IE5gwuIqoqGNIAaBiNCdcA]
CSeq: 32079 INVITE
Route: sip:ssbctrunk-us.8x8.com:5500;lr
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: Asterisk PBX 20.5.2
Content-Type: application/sdp
Content-Length: 239

I believe you can also use \;hide

I don’t know if that is a PJSIP special or something in an RFC, but I gather it suppresses the route header. Howerver, with loose routing a Route header that matches the proxy should get deleted.

PS Do not start a new topic for continuations of the same basic issue.

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