[OpenSIPS-Users] Invalid dialog

[James Seer]

Hello,

I am experiencing strange behaviour from one specific provider.
80% of calls work properly, some of them fall into the validate_dialog
condition, returncode = -1
i've checked INVITE and ACK message , i don't see anything that could cause
that.
fix_dialog_route seems to fix the problem but I would love to understand
what is causing it.
i noticed that with other providers when the dialog is not valid there is
an ERROR message in the log files
"ERROR:dialog:dlg_validate_dialog: failed to validate remote contact"
with this one , there is no error.
I compared the call with another working one from the same provider , I see
no difference.
Here are the initial invite, the 200 ok from opensips to provider, and the
ACK that does not match during dialog validation.

proxyfqdn.example:  my opensips FQDN
pro.vid.der.ip : provider ip address
my.pbx.behind.opensips.ip : my pbx behind opensips (asterisk box)

INITIAL INVITE :

INVITE sip:0100000010 at proxyfqdn.example SIP/2.0
Via: SIP/2.0/UDP pro.vid.der.ip:5060;branch=z9hG4bK695c5157
Max-Forwards: 70
From: "anonymous" <sip:6134673 at pro.vid.der.ip>;tag=as6a0e2632
To: <sip:0100000010 at proxyfqdn.example>
Contact: <sip:6134673 at pro.vid.der.ip:5060>
Call-ID: 0211c8466462c5e942f58e7479e81c08 at pro.vid.der.ip:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.3.0
Date: Mon, 17 Apr 2023 18:52:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
X-ADDTIONAL-UUID: <p(<
Privacy: id
P-Asserted-Identity: <sip:+XXXXXXXXX at pro.vid.der.ip:5060>
XSHDATAID: T1798
Diversion: <sip:+XXXXXXXXXX at pro.vid.der.ip>;reason=unconditional
Content-Type: application/sdp
Content-Length: 250

v=0
o=root 665874795 665874795 IN IP4 pro.vid.der.ip
s=Asterisk PBX 18.3.0
c=IN IP4 pro.vid.der.ip
t=0 0
m=audio 7162 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


--------------------------------------

200 OK (Proxy to provider)

SIP/2.0 200 OK
Via: SIP/2.0/UDP pro.vid.der.ip:5060;branch=z9hG4bK695c5157
Record-Route:
<sip:pro.xy.ip.address:5060;lr;ftag=as6a0e2632;did=1bc.b66272e6>
From: "anonymous" <sip:6134673 at pro.vid.der.ip>;tag=as6a0e2632
To: <sip:0100000010 at proxyfqdn.example>;tag=as2b2572fa
Call-ID: 0211c8466462c5e942f58e7479e81c08 at pro.vid.der.ip:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.11.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+XX100000010 at my.pbx.behind.opensips.ip:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 237

v=0
o=root 1475118129 1475118129 IN IP4 my.pbx.behind.opensips.ip
s=MYMEDIASERVER
c=IN IP4 my.pbx.behind.opensips.ip
t=0 0
m=audio 17486 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

--------------------------------------------------------

ACK that cause dialog invalidation :


ACK sip:+XX100000010 at my.pbx.behind.opensips.ip:5060 SIP/2.0
Via: SIP/2.0/UDP pro.vid.der.ip:5060;branch=z9hG4bK11c1df1f
Route: <sip:pro.xy.ip.address:5060;lr;ftag=as6a0e2632;did=1bc.b66272e6>
Max-Forwards: 70
From: "anonymous" <sip:6134673 at pro.vid.der.ip>;tag=as6a0e2632
To: <sip:0100000010 at proxyfqdn.example>;tag=as2b2572fa
Contact: <sip:6134673 at pro.vid.der.ip:5060>
Call-ID: 0211c8466462c5e942f58e7479e81c08 at pro.vid.der.ip:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.3.0
Content-Length: 0
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