Hi Mathias,

I am new to this and have been playing about with Kamailio & Asterisk. I also use the dispatcher to send calls to Asterisk from Kamailio. I am not seeing exactly what you are doing, but what I have done is 2 things:

1. In the "auth" part of Kamailio, I have used IP address to detect messages coming FROM Asterisk, as I am trusting calls from these nodes and they are isolated from external injection etc. If they arrive from Asterisk I use a 'flag' to mark the call (as all calls coming from my Asterisk were generally sent there by Kamailio in the first place).

2. Also, when I send then to Asterisk I add extra info in the "Route", such as 'media=yes;' (I call my Asterisk boxes as media handlers)

In my 'withindlg' script, I can then do things like:

if($(hdr(Route){param.value,media}) == "yes") {                         route(MEDIA);                 } else {

route(RELAY);                 }                 exit;

This is just a small part, but maybe that principle may be of some help?

Steve

On 12/03/2019 07:48, YASIN CANER wrote:

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/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:004112345@10.40.8.104:5060 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

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*From:* sr-users sr-users-bounces@lists.kamailio.org on behalf of Schneuwly Mathias RUAG Mathias.Schneuwly@ruag.com *Sent:* Tuesday, March 12, 2019 10:06 AM *To:* sr-users@lists.kamailio.org *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:proxydevice@10.40.8.104;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC         To: sip:004112345@10.40.8.104;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:004112345@10.40.8.104:5060         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:proxydevice@10.40.8.104;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC         To: sip:004112345@10.40.8.104;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:004112345@10.40.8.104:5060         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:004112345@10.40.8.104:5060 |

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:004112345@10.40.8.104:5060 SIP/2.0         Method: ACK         Request-URI: sip:004112345@10.40.8.104:5060         [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:proxydevice@10.40.8.104;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC         To: sip:004112345@10.40.8.104;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:proxydevice@10.40.8.104;tag=NCP2N5E7OZuppblpkCCNZziEmpiTRsQC         To: sip:004112345@10.40.8.104;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:004112345@10.40.6.214:49316;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:004112345@10.40.8.106:5060 |

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:004112345@10.40.8.106:5060 |

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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