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@192.168.20.1mailto: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.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: miconda@gmail.commailto:miconda@gmail.com; Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto: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.commailto:miconda@gmail.com> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org>; Martin W Woscek <mwoscek@mitre.orgmailto: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: Hello all,

SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone]

But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio

Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF.

Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE.

Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible.

Thanks, _Martin

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