Hi,
looking at the trace, it seems to me that the 407 is not sent by Kamailio
but by the FS1 server ("User-Agent: IMX" in the 407, as for the 1st 407
sent by FS2), and that you are probably missing in the trace the messages
between Kamailio and FS1.
If this is the case you could either:
- disable the authentication in freeswitch if you don't need it
- store the custom header when receiving the 302 and use it to set $du upon
receiving the 2nd INVITE
For the latter you can have several solutions, I suggest you to have a a
look at the htable module. For example you could use it to store the key
USERID::CALLID with the value of your custom header and retrieve it when
you receive the authenticated INVITE from the client. Something like
sht(TABLE_NAME=>$fU::$ci) = $avp(imx-redirect);
to store it, and then
$avp(imx-redirect) = sht(TABLE_NAME=>$fU::$ci);
to retrieve it.
Hope this helps.
CHeers,
Federico
On Tue, Sep 16, 2014 at 3:28 AM, Nicholas Gill <nick(a)etellicom.com> wrote:
On 15/09/14 16:03, Federico Cabiddu wrote:
What happens if you call t_relay() after setting
$du?
Same result. Kamailio sends 407 to caller, which then creates a new INVITE.
I suspect my configuration doesn't reflect the intent, so I'm adding it
below along with other particulars of the test environment.
Is there a specific way to cause kamailio to relay the INVITE to a
particular server based on a dynamic variable?
Cheers,
-nick
192.168.20.215 - Caller
192.168.20.214:5060 Kamailio (
node1.etellicom.com)
192.168.20.214:5080 FS1 (
node1.etellicom.com)
192.168.20.198:5080 FS2 (
node2.etellicom.com)
Kamailio:
version: kamailio 4.1.5 (x86_64/linux) 2b3d4b
Configuration diff from default:
*** kamailio.cfg.default 2014-08-29 15:07:42.336496172 +1000
--- kamailio.cfg 2014-09-16 10:43:50.199400261 +1000
*************** loadmodule "xmlrpc.so"
*** 283,288 ****
--- 283,296 ----
loadmodule "debugger.so"
#!endif
+ loadmodule "dispatcher.so"
+ modparam("dispatcher", "flags", 2)
+ modparam("dispatcher", "list_file",
"/root/dispatcher.list")
+ modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
+ modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
+ modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
+ #modparam("dispatcher", "ds_ping_interval", 60)
+
# ----------------- setting module-specific parameters ---------------
*************** request_route {
*** 491,496 ****
--- 499,509 ----
setflag(FLT_ACC); # do accounting
}
+ if(!ds_is_from_list("1"))
+ {
+ route(DISPATCH);
+ }
+
# dispatch requests to foreign domains
route(SIPOUT);
*************** request_route {
*** 516,521 ****
--- 529,576 ----
route(LOCATION);
}
+ # Dispatch requests
+ route[DISPATCH] {
+
+ # Hash over to uri for test purposes
+ if(!ds_select_dst("1", "2"))
+ {
+ send_reply("404", "No destination");
+ exit;
+ }
+ xlog("L_DBG", "--- SCRIPT: going to <$ru> via
<$du>\n");
+ t_on_failure("RTF_DISPATCH");
+ route(RELAY);
+ exit;
+ }
+
+ # Dispatch failure route
+ failure_route[RTF_DISPATCH] {
+ if (t_is_canceled()) {
+ exit;
+ }
+ # next DST - only for 500 or local timeout
+ if (t_check_status("500")
+ or (t_branch_timeout() and !t_branch_replied()))
+ {
+ if(ds_next_dst())
+ {
+ t_on_failure("RTF_DISPATCH");
+ route(RELAY);
+ exit;
+ }
+ }
+
+ if ($avp(imx-redirect) != $null)
+ {
+ $du = $avp(imx-redirect);
+ xlog("SCRIPT: REDIRECT: going to <$ru> via <$du>\n");
+ # route(RELAY);
+ t_relay();
+ exit;
+ }
+
+ }
route[RELAY] {
*************** onreply_route[MANAGE_REPLY] {
*** 911,916 ****
--- 966,977 ----
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
+
+ if (t_check_status("3[0-9][0-9]") and
is_present_hf("X-imx-redirect"
))
+ {
+ $avp(imx-redirect) = $hdr(X-imx-redirect);
+ xlog("SCRIPT: REDIRECT, Store avp for redirect:
$avp(imx-redirect)\n");
+ }
}
# manage failure routing cases
SIP trace of call intended to be redirected to a specific server.
Scenario starts with normal INVITE, 100, proxy auth from backend server,
INVITE with auth.
When FS2 receives the authenticated invite, it responds with 302 and
indicates via the X-imx-redirect header to redirect the request to FS1.
Then I want to send the INVITE to the given server by setting $du and
calling t_relay() but my current configuration seems to cause kamailio to
respond to the caller with 407 instead.
10:46:27.361991 IP 192.168.20.215.5062 > 192.168.20.214.5060
INVITE sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 INVITE
Contact: <sip:1001@192.168.20.215:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T38G 38.70.0.115
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 310
SDP
10:46:27.362567 IP 192.168.20.214.5060 > 192.168.20.215.5062
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
10:46:27.362745 IP 192.168.20.214.5060 > 192.168.20.198.5080
INVITE sip:*591000@test.etellicom.com SIP/2.0
Record-Route: <sip:192.168.20.214;lr=on>
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bKa4a8.
9a165b4901cf1602b3146c7f0bf0a191.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 INVITE
Contact: <sip:1001@192.168.20.215:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: Yealink SIP-T38G 38.70.0.115
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 310
SDP
10:46:27.383907 IP 192.168.20.198.5080 > 192.168.20.214.5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bKa4a8.
9a165b4901cf1602b3146c7f0bf0a191.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
Record-Route: <sip:192.168.20.214;lr=on>
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 INVITE
User-Agent: IMX
Content-Length: 0
10:46:27.553363 IP 192.168.20.198.5080 > 192.168.20.214.5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bKa4a8.
9a165b4901cf1602b3146c7f0bf0a191.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=Ke2rvve07SK3N
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 INVITE
User-Agent: IMX
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Proxy-Authenticate: Digest realm="test.etellicom.com", nonce="xxx",
algorithm=MD5, qop="auth"
Content-Length: 0
10:46:27.553574 IP 192.168.20.214.5060 > 192.168.20.198.5080
ACK sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bKa4a8.
9a165b4901cf1602b3146c7f0bf0a191.0
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=Ke2rvve07SK3N
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0
10:46:27.553718 IP 192.168.20.214.5060 > 192.168.20.215.5062
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=Ke2rvve07SK3N
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 INVITE
User-Agent: IMX
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Proxy-Authenticate: Digest realm="test.etellicom.com", nonce="xxx",
algorithm=MD5, qop="auth"
Content-Length: 0
10:46:27.592662 IP 192.168.20.215.5062 > 192.168.20.214.5060
ACK sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK1598556947
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=Ke2rvve07SK3N
Call-ID: 1766438446(a)192.168.20.215
CSeq: 1 ACK
Content-Length: 0
10:46:27.630142 IP 192.168.20.215.5062 > 192.168.20.214.5060
INVITE sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 INVITE
Contact: <sip:1001@192.168.20.215:5062>
Proxy-Authorization: Digest username="1001",
realm="test.etellicom.com",
nonce="xxx", uri="sip:*591000@test.etellicom.com",
response="xxx",
algorithm=MD5, cnonce="xxx", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T38G 38.70.0.115
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 310
SDP
10:46:27.630721 IP 192.168.20.214.5060 > 192.168.20.215.5062
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
10:46:27.630896 IP 192.168.20.214.5060 > 192.168.20.198.5080
INVITE sip:*591000@test.etellicom.com SIP/2.0
Record-Route: <sip:192.168.20.214;lr=on>
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bK74a8.
d3d3ad8bb9b1c73c12e16cdab447472e.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 INVITE
Contact: <sip:1001@192.168.20.215:5062>
Proxy-Authorization: Digest username="1001",
realm="test.etellicom.com",
nonce="xxx", uri="sip:*591000@test.etellicom.com",
response="xxx",
algorithm=MD5, cnonce="xxx", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: Yealink SIP-T38G 38.70.0.115
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 310
SDP
10:46:27.631731 IP 192.168.20.198.5080 > 192.168.20.214.5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bK74a8.
d3d3ad8bb9b1c73c12e16cdab447472e.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
Record-Route: <sip:192.168.20.214;lr=on>
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 INVITE
User-Agent: IMX
Content-Length: 0
10:46:29.187901 IP 192.168.20.198.5080 > 192.168.20.214.5060
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bK74a8.
d3d3ad8bb9b1c73c12e16cdab447472e.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
Max-Forwards: 69
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=mQUHyQZ3429NH
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 INVITE
Contact: "unknown" <sip:node1.etellicom.com>
User-Agent: IMX
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Reason: Q.850;cause=31;text="NORMAL_UNSPECIFIED"
Content-Length: 0
X-imx-redirect: sip:192.168.20.214:5080
10:46:29.188327 IP 192.168.20.214.5060 > 192.168.20.198.5080
ACK sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bK74a8.
d3d3ad8bb9b1c73c12e16cdab447472e.0
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=mQUHyQZ3429NH
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 ACK
Max-Forwards: 69
Content-Length: 0
7(17488) ERROR: <script>: SCRIPT: REDIRECT, Store avp for redirect: sip:
192.168.20.214:5080
7(17488) ERROR: <script>: SCRIPT: REDIRECT: going to <sip:*
591000(a)test.etellicom.com> via <sip:192.168.20.214:5080>
10:46:29.432573 IP 192.168.20.214.5060 > 192.168.20.215.5062
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=ZS5Q6Xem0Umaj
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 INVITE
User-Agent: IMX
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Proxy-Authenticate: Digest realm="test.etellicom.com",
nonce="eac7076b-4c13-4492-838c-14039cc2beb6", stale=true, algorithm=MD5,
qop="auth"
Content-Length: 0
10:46:29.452446 IP 192.168.20.215.5062 > 192.168.20.214.5060
ACK sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK416062902
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>;tag=ZS5Q6Xem0Umaj
Call-ID: 1766438446(a)192.168.20.215
CSeq: 2 ACK
Content-Length: 0
10:46:29.490390 IP 192.168.20.215.5062 > 192.168.20.214.5060
INVITE sip:*591000@test.etellicom.com SIP/2.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK2079130033
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 3 INVITE
Contact: <sip:1001@192.168.20.215:5062>
Proxy-Authorization: Digest username="1001",
realm="test.etellicom.com",
nonce="xxx", uri="sip:*591000@test.etellicom.com",
response="xxx",
algorithm=MD5, cnonce="xxx", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T38G 38.70.0.115
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 310
SDP
10:46:29.490694 IP 192.168.20.214.5060 > 192.168.20.215.5062
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK2079130033
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 3 INVITE
Server: kamailio (4.1.5 (x86_64/linux))
Content-Length: 0
10:46:29.490782 IP 192.168.20.214.5060 > 192.168.20.198.5080
INVITE sip:*591000@test.etellicom.com SIP/2.0
Record-Route: <sip:192.168.20.214;lr=on>
Via: SIP/2.0/UDP 192.168.20.214;branch=z9hG4bK84a8.
289b7cd802ed6efd2f25721d6b8f29d6.0
Via: SIP/2.0/UDP 192.168.20.215:5062;branch=z9hG4bK2079130033
From: "1001" <sip:1001@test.etellicom.com>;tag=1860002040
To: <sip:*591000@test.etellicom.com>
Call-ID: 1766438446(a)192.168.20.215
CSeq: 3 INVITE
Contact: <sip:1001@192.168.20.215:5062>
Proxy-Authorization: Digest username="1001",
realm="test.etellicom.com",
nonce="xxx", uri="sip:*591000@test.etellicom.com",
response="xxx",
algorithm=MD5, cnonce="xxx", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: Yealink SIP-T38G 38.70.0.115
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 310
SDP
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