Hello,
Just wanted to update what fixed our issue.
The “contact_user” which is “asterisk” that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation).
However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the correct
flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk.
Anyway, its always good to end Friday on a high note.
Thanks,
_Martin
From: sr-users <sr-users-bounces@lists.kamailio.org>
On Behalf Of Martin W Woscek
Sent: Wednesday, June 24, 2020 1:32 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>; miconda@gmail.com
Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct.
The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF
should send this INVITE to the UE (703222888@192.168.20.1)
No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type
Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host
Port Info
1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt
sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99
7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE
sip:7032228888@192.168.20.1:63871;transport=udp |
Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0
Linux cooked capture
Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128
User Datagram Protocol, Src Port: 6060, Dst Port: 4060
Session Initiation Protocol (INVITE)
Request-Line: INVITE
sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0
Method: INVITE
Request-URI:
sip:7032228888@192.168.20.1:63871;transport=udp
Request-URI User Part: 7032228888
Request-URI Host Part: 192.168.20.1
Request-URI Host Port: 63871
[Resent Packet: False]
Message Header
Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1>
Route: <sip:term@pcscf522.irisims.org;lr>
Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0
Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1
Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335
From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d
To: <sip:7032228888@ims522.irisims.org>
Contact: <sip:asterisk@192.168.20.131:5060>
Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660
[Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660]
CSeq: 19314 INVITE
Route: <sip:ims522.irisims.org:5060;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 68
User-Agent: Asterisk PBX 16.6.0
Content-Type: application/sdp
Content-Length: 1121
Message Body
From: sr-users <sr-users-bounces@lists.kamailio.org>
On Behalf Of Martin W Woscek
Sent: Wednesday, June 24, 2020 11:21 AM
To: miconda@gmail.com; Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending
SIP:200 OK messages to Asterisk (to tag)
Hi,
The issue is the P should be sending to the UE not back to the I like the failed flow:
Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone]
The successful call should look like:
Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE]
Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work?
The reverse calls work just fine.
Thanks,
_Martin
From: Daniel-Constantin Mierla <miconda@gmail.com>
Sent: Wednesday, June 24, 2020 4:53 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>; Martin W Woscek <mwoscek@mitre.org>
Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending
SIP:200 OK messages to Asterisk (to tag)
Hello,
Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag).
Cheers,
Daniel
On 19.06.20 21:19, Martin W Woscek wrote: