Hi James,
Here are my listen directives:
listen=udp:192.168.1.2:5060
listen=tcp: 192.168.1.2:5060
knowing that:
Kamailio IP: 192.168.1.2
SBC IP: 192.168.1.1
Caller IP: 192.168.1.5
-----Original Message-----
From: Ali Taher via sr-users <sr-users(a)lists.kamailio.org>
Sent: Thursday, November 2, 2023 1:42 PM
To: James Browne <james(a)frideo.com>om>; Kamailio (SER) - Users Mailing List
<sr-users(a)lists.kamailio.org>
Cc: Ali Taher <ataher(a)vanrise.com>
Subject: [SR-Users] Re: SIP Proxy : BYE next hop
Hi James,
Yes I'm using loose_route() and it's returning 1, but BYE message is not routed
anywhere.
Frame 6213: 673 bytes on wire (5384 bits), 673 bytes captured (5384 bits) Ethernet II,
Src: Cisco_f8:19:ff (00:22:bd:f8:19:ff), Dst: VMware_bf:27:0b (00:50:56:bf:27:0b) Internet
Protocol Version 4, Src: 192.168.1.5, Dst: 192.168.1.2 User Datagram Protocol, Src Port:
16368, Dst Port: 5060 Session Initiation Protocol (BYE)
Request-Line: BYE sip:192.168.1.1:5060;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP
192.168.1.5:16368;branch=z9hG4bK-d87543-c147891e677a1f19-1--d87543-;rport
Max-Forwards: 70
Route: <sip:192.168.1.2;lr;ftag=7d4a7c3c>
Route: <sip:192.168.1.1:5060;ipcs-line=43354518;lr;transport=udp>
Contact: <sip:131@192.168.1.5:16368>
To: "121"<sip:121@192.168.1.1>;tag=gtyrzghr-CC-1047-OFC-39
From: "69079514"<sip:69079514@192.168.1.2>;tag=7d4a7c3c
Call-ID: 085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.
[Generated Call-ID:
085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.]
CSeq: 2 BYE
User-Agent: eyeBeam release 1003s stamp 31159
Reason: SIP;description="User Hung Up"
Content-Length: 0
Here is routing block related to BYE:
if (is_method("BYE")) {
xlog("L_INFO","INSIDE BYE");
$var(loose) = loose_route();
xlog("L_INFO","LOOSE : $var(loose)");
}
Regards,
-----Original Message-----
From: James Browne <james(a)frideo.com>
Sent: Thursday, November 2, 2023 12:31 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Cc: Ali Taher <ataher(a)vanrise.com>
Subject: Re: [SR-Users] SIP Proxy : BYE next hop
Hello
Did you call the loose_route() function for the BYE? It should just work; that's one
of the basics of how kamailio works, so I'll be very surprised if there's somehow
a bug in kamailio.
If that does not work, perhaps send a list of the listen directives in the kamailio.cfg,
along with the headers of the two SIP BYE messages (incoming and outgoing) and I'll
see what I can figure out for you.
James
On Thu, 2 Nov 2023 at 09:35, Ali Taher via sr-users <sr-users(a)lists.kamailio.org>
wrote:
Hello all,
I’m using Kamailio as SIP proxy between session manager and SBC.
I came across an issue in BYE request where the SBC sent in the header
list of Routes where first route is Kamailio IP, and the second is SM
IP, in addition to other routes (last route is the caller party)
What Kamailio did is to route the BYE to the address set in the Request-URI, which is the
caller party.
What should be done instead is that Kamailio should remove the Route header including its
own IP, and route the BYE to the next hop which is in this case the SM IP.
How to accomplish that?
Following is the Route header in the BYE request coming from SBC:
[truncated]Route:
<sip:KAMAILIO_IP;lr;ftag=xxxxxxxxxxxxxxxxxxxxxxx>,<sip:SM1@SM_IP;trans
port=udp;lr;av-asset-uid=xxxxxxxxxx>,<sip:127.0.0.2:15060;transport=ud
p;ibmsid=local.xxxxxxxxxxxx_xxxxxxxxx_xxxxxxxxx;lr;ibmdrr>,sips:127.0.
0.2:15061;ibmsid=local.xxxxxxxxxx_xxxxxxxxxx_xxxxxxxx;lr;ibmdrr,
sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xxxxxx,
sip:CALLER_PARTY_IP:5061;transport=tls;lr
Regards,
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