Hello, It is hard to understand with these logs but it looks like loose_route issiue.
in WITHINDLG route (sample script) there is a loose_route function that function set route header uri to Request uri if it is related with kamailio as shown below logs. so , kamailio send ACK message to itself that shown at second. it is specified in RFC 3261 <https://tools.ietf.org/html/rfc3515> https://tools.ietf.org/html/rfc3261#section-16.12 Best Regards Yasin CANER. Internet Protocol Version 4, Src: 10.40.6.188, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 56960, Dst Port: 5070 Session Initiation Protocol (ACK) Request-Line: ACK sip:[email protected]:5060 SIP/2.0 Route: <sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61> Internet Protocol Version 4, Src: 10.40.8.104, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 5070, Dst Port: 5070 Session Initiation Protocol (ACK) Request-Line: ACK sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61 SIP/2.0 ________________________________ From: sr-users <[email protected]> on behalf of Schneuwly Mathias RUAG <[email protected]> Sent: Tuesday, March 12, 2019 10:06 AM To: [email protected] Subject: Re: [SR-Users] ACK not forwarding to Asterisk Dear Kamailio mailing list I've a problem with a simple Kamailio-Asterisk setup. I tried to find a solution and found several posts, but was not able to fix my problem! I'm using the following setup: - SIP device registered to Kamailio (IP 10.40.6.188) - Kamailio (port 5070) and Asterisk (port 5060) on the same host (IP 10.40.8.104) - I tried to forward calls from Kamailio to Asterisk with with standard PSTN call routing and dispatcher module, but the result was the same - Asterisk has a trunk to Kamailio - SIP device registered to Asterisk (IP 10.40.6.214) Calls from the Asterisk phone towards the Kamailio phone work without any issue. Calls from the Kamailio phone towards the Asterisk phone are successfully established and the voice works in both directions. The problem here is, that the Kamailio phone sends an ACK upon 200 OK, but the ACK is not forwarded to Asterisk. After 6 seconds, the call is terminated. If I remove record_route() or add a second IP to the same interface for Asterisk (10.40.8.106), then the call signaling works without any issue. Does anyone have an idea what I'm doing wrong here? I'm using more or less a standard Kamailio config file without rtpproxy. Please let me know if I shall post the full config or just some relevant snippets. Best regards Mathias Following are two snippets of the relevant ACK. The first shows the situation where the ACK was not forwarded to Asterisk and the second with a different IP for Asterisk works. No. Time Source Destination Protocol Length Info 752 17.269019 10.40.8.104 10.40.8.104 SIP/SDP 1135 Status: 200 OK | Frame 752: 1135 bytes on wire (9080 bits), 1135 bytes captured (9080 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.104, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 5060, Dst Port: 5070 Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] [Request Frame: 304] [Response Time (ms): 2093] Message Header Via: SIP/2.0/UDP 10.40.8.104:5070;branch=z9hG4bKda4f.48918e0ef9372ba1a38655eb93742e46.0;received=10.40.8.104 Via: SIP/2.0/UDP 10.40.6.188:56960;received=10.40.6.188;rport=56960;branch=z9hG4bKPjVtFCvs26liH-aJoGn8CaLg4kWqml5LH. Record-Route: <sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61> From: <sip:[email protected]>;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC To: <sip:[email protected]>;tag=as5e5c287c Call-ID: a6T0GAwBL69m5gsc215EgXi61Oorwxik CSeq: 10909 INVITE Server: Asterisk PBX 12.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp Require: timer Content-Length: 271 Message Body No. Time Source Destination Protocol Length Info 764 17.270668 10.40.8.104 10.40.6.188 SIP/SDP 1025 Status: 200 OK | Frame 764: 1025 bytes on wire (8200 bits), 1025 bytes captured (8200 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.104, Dst: 10.40.6.188 User Datagram Protocol, Src Port: 5070, Dst Port: 56960 Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] [Request Frame: 259] [Response Time (ms): 2115] Message Header Via: SIP/2.0/UDP 10.40.6.188:56960;received=10.40.6.188;rport=56960;branch=z9hG4bKPjVtFCvs26liH-aJoGn8CaLg4kWqml5LH. Record-Route: <sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61> From: <sip:[email protected]>;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC To: <sip:[email protected]>;tag=as5e5c287c Call-ID: a6T0GAwBL69m5gsc215EgXi61Oorwxik CSeq: 10909 INVITE Server: Asterisk PBX 12.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp Require: timer Content-Length: 271 Message Body No. Time Source Destination Protocol Length Info 786 17.402987 10.40.6.188 10.40.8.104 SIP 486 Request: ACK sip:[email protected]:5060 | Frame 786: 486 bytes on wire (3888 bits), 486 bytes captured (3888 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.6.188, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 56960, Dst Port: 5070 Session Initiation Protocol (ACK) Request-Line: ACK sip:[email protected]:5060 SIP/2.0 Method: ACK Request-URI: sip:[email protected]:5060 [Resent Packet: False] [Request Frame: 259] [Response Time (ms): 2247] Message Header Via: SIP/2.0/UDP 10.40.6.188:56960;rport;branch=z9hG4bKPjM2GQWjVep2XhpLxJD-cxNh75Qtep-sN3 Max-Forwards: 70 From: <sip:[email protected]>;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC To: <sip:[email protected]>;tag=as5e5c287c Call-ID: a6T0GAwBL69m5gsc215EgXi61Oorwxik CSeq: 10909 ACK Route: <sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61> Content-Length: 0 No. Time Source Destination Protocol Length Info 787 17.404480 10.40.8.104 10.40.8.104 SIP 561 Request: ACK sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61 | Frame 787: 561 bytes on wire (4488 bits), 561 bytes captured (4488 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.104, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 5070, Dst Port: 5070 Session Initiation Protocol (ACK) Request-Line: ACK sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61 SIP/2.0 Method: ACK Request-URI: sip:10.40.8.104:5070;lr;ftag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC;did=17a.4f61 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 10.40.8.104:5070;branch=z9hG4bKda4f.f914ac9def098ece9a50d53389574f85.0 Via: SIP/2.0/UDP 10.40.6.188:56960;received=10.40.6.188;rport=56960;branch=z9hG4bKPjM2GQWjVep2XhpLxJD-cxNh75Qtep-sN3 Max-Forwards: 69 From: <sip:[email protected]>;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC To: <sip:[email protected]>;tag=as5e5c287c Call-ID: a6T0GAwBL69m5gsc215EgXi61Oorwxik CSeq: 10909 ACK Content-Length: 0 In this case I used a second IP on the same interface for Asterisk and this call works without any issue: No. Time Source Destination Protocol Length Info 939 8.849431 10.40.8.106 10.40.6.214 SIP 473 Request: ACK sip:[email protected]:49316;ob | Frame 939: 473 bytes on wire (3784 bits), 473 bytes captured (3784 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.106, Dst: 10.40.6.214 User Datagram Protocol, Src Port: 5060, Dst Port: 49316 Session Initiation Protocol (ACK) No. Time Source Destination Protocol Length Info 946 8.850293 10.40.8.106 10.40.8.104 SIP/SDP 1136 Status: 200 OK | Frame 946: 1136 bytes on wire (9088 bits), 1136 bytes captured (9088 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.106, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 5060, Dst Port: 5070 Session Initiation Protocol (200) No. Time Source Destination Protocol Length Info 956 8.851703 10.40.8.104 10.40.6.188 SIP/SDP 1026 Status: 200 OK | Frame 956: 1026 bytes on wire (8208 bits), 1026 bytes captured (8208 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.104, Dst: 10.40.6.188 User Datagram Protocol, Src Port: 5070, Dst Port: 60223 Session Initiation Protocol (200) No. Time Source Destination Protocol Length Info 989 9.095859 10.40.6.188 10.40.8.104 SIP 485 Request: ACK sip:[email protected]:5060 | Frame 989: 485 bytes on wire (3880 bits), 485 bytes captured (3880 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.6.188, Dst: 10.40.8.104 User Datagram Protocol, Src Port: 60223, Dst Port: 5070 Session Initiation Protocol (ACK) No. Time Source Destination Protocol Length Info 991 9.097228 10.40.8.104 10.40.8.106 SIP 516 Request: ACK sip:[email protected]:5060 | Frame 991: 516 bytes on wire (4128 bits), 516 bytes captured (4128 bits) Linux cooked capture Internet Protocol Version 4, Src: 10.40.8.104, Dst: 10.40.8.106 User Datagram Protocol, Src Port: 5070, Dst Port: 5060 Session Initiation Protocol (ACK)
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