[Kamailio-Users] add_path_received and OBP problems
Klaus Darilion
klaus.mailinglists at pernau.at
Mon Sep 28 08:42:51 CEST 2009
Florian Meister schrieb:
> Hi,
>
> as thought invite messages are routed now correctly to the public IP
> of the UAC.
>
> Now I have another problem. ACK and BYE messages have the same
> problem. The loadbalancer(s) are trying to send the messages to the
> private IP, which of course does not work.
>
> I thought at least the BYE message is an in-dialog-request, and
> therefore it should work ? Do you have an idea how to route this kind
> of messages correctly through the LB to the UAC's ?
How do you route in-dialog messages? LB and proxy/registrar should use
record-route on initial request to stay in the proxy-chain. LB and
proxy/registrar should route in-dialog requests with loose_route() and
t_relay().
Also, on the LB you have to do the NAT traversal for the clients: i.e.
fix_nated_contact() on messages from the client. If the LB is multihomed
and uses double-record-routing I think you could the same mechanism like
path (adding "received" to the record-route header) too.
regards
klaus
>
> best regards,
>
> Florian
>
>>> The request sent out by the LB still has the Route header. This
>>> indicates that either: - the load balancer does not detect the
>>> domain/IP in the Route
>> header
>>> as an internal one (alias=....) - the loose_route function is not
>>> called in the LB.
>>>
>> Many thanks ! I did not call loose_route in the LB, I think that
>> will solve it.
>>
>> Best regards,
>>
>> Florian
>>
>>
>>> Florian Meister schrieb:
>>>> Hi,
>>>>
>>>> I'm trying to set up a HA-OBP for both natted and non-natted
>>>> users.
>>>>
>>>> Atm my setup looks like that:
>>>>
>>>>
>>>> [UAC] <---> (NAT) <---> [LB] <---> [REG + Proxy] <---> [PSTN
>>>> GW]
>>>>
>>>> At the loadbalancer I use add_path and add_path_received to add
>>>> a
>>> path-header to the request sent to the Registrar. (Of course only
>>> for register requests). I use a few parameters for telling the
>>> registrar module to use the path header. When looking in the
>>> mysql-db I can see the correct value in the path-field in the
>>> location-table.
>>>> When getting an invite from the PSTN gateway for one of the
>>>> UAC's
>> the
>>> Proxy is adding a route-header according to the path-value in the
>>> mysql. That seems also correct. Now to my problem:
>>>> The loadbalancer is sending out the INVITE to the private-ip,
>>>> not
>> to
>>> the public-nat-ip of the UAC. It seems that the
>>> received-parameter in the Route-header is somehow ignored. I am
>>> using the use_received parameter in the loadbalancer, but it does
>>> not work.
>>>> In the following you can find the SIP trace (made on the
>>> loadbalancer) and the loadbalancer config, since I think there is
>>> the problem.
>>>> Thanks,
>>>>
>>>> Florian
>>>>
>>>>
>>>> Here is the SIP-trace of the INVITE:
>>>>
>>>> 09:49:07.785294 IP [Proxy+Registrar].5060 >
>>>> [Loadbalancer].5060:
>> SIP,
>>> length: 1101
>>>> E..i.. at .?.Yb..,F..,+.....U.BINVITE sip:435572949012@[PRIVATE IP
>>>> of
>>> UAC];user=phone SIP/2.0
>>>> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via:
>>>> SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
>>> UAC]:64224">
>>>> Max-Forwards: 69 From: "4366480588134" <sip:4366480588134@[PSTN
>>>>
>>> Gateway]>;tag=as33339e2e
>>>> To: <sip:+435572949012@[Proxy+Registrar]> Contact:
>>>> <sip:4366480588134@[PSTN Gateway]> Call-ID:
>>>> 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102
>>>> INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009
>>>> 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>>> SUBSCRIBE, NOTIFY,
>>> INFO
>>>> Supported: replaces, timer Content-Type: application/sdp
>>>> Content-Length: 312
>>>>
>>>> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
>>>> s=Asterisk PBX 1.6.1.5 c=IN IP4 [PSTN Gateway] t=0 0 m=audio
>>>> 19178 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3
>>>> GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 09:49:07.786459 IP [Loadbalancer].5060 >
>>>> [Proxy+Registrar].5060:
>> SIP,
>>> length: 445
>>>> E..... at .@.Z...,+..,F........SIP/2.0 100 Giving a try Via:
>>>> SIP/2.0/UDP
>> [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
>>
>>> oxy+Registrar]
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> From: "4366480588134" <sip:4366480588134@[PSTN
>>> Gateway]>;tag=as33339e2e
>>>> To: <sip:+435572949012@[Proxy+Registrar]> Call-ID:
>>>> 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102
>>>> INVITE Server: Kamailio (1.5.1-notls (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> 09:49:07.786608 IP [Loadbalancer].5060 > [PRIVATE IP of
>>>> UAC].5060:
>>> SIP, length: 1307
>>>> E..7.. at .@.. at ..,+.........#>.INVITE sip:435572949012@[PRIVATE IP
>>>> of
>>> UAC];user=phone SIP/2.0
>>>> Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
>>>> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via:
>>>> SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via:
>>>> SIP/2.0/UDP
>> [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
>>
>>> d51.bf3830b5.0
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
>>> UAC]:64224">
>>>> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN
>>>>
>>> Gateway]>;tag=as33339e2e
>>>> To: <sip:+435572949012@[Proxy+Registrar]> Contact:
>>>> <sip:4366480588134@[PSTN Gateway]> Call-ID:
>>>> 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102
>>>> INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009
>>>> 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>>> SUBSCRIBE, NOTIFY,
>>> INFO
>>>> Supported: replaces, timer Content-Type: application/sdp
>>>> Content-Length: 330 Path: <sip:[Loadbalancer];lr>
>>>>
>>>> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
>>>> s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio
>>>> 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3
>>>> GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
>>>> a=sendrecv a=nortpproxy:yes
>>>>
>>>> 09:49:08.258116 IP [Loadbalancer].5060 > [PRIVATE IP of
>>>> UAC].5060:
>>> SIP, length: 1307
>>>> E..7.. at .@.. at ..,+.........#>.INVITE sip:435572949012@[PRIVATE IP
>>>> of
>>> UAC];user=phone SIP/2.0
>>>> Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
>>>> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via:
>>>> SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via:
>>>> SIP/2.0/UDP
>> [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
>>
>>> d51.bf3830b5.0
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
>>> UAC]:64224">
>>>> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN
>>>>
>>> Gateway]>;tag=as33339e2e
>>>> To: <sip:+435572949012@[Proxy+Registrar]> Contact:
>>>> <sip:4366480588134@[PSTN Gateway]> Call-ID:
>>>> 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102
>>>> INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009
>>>> 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>>> SUBSCRIBE, NOTIFY,
>>> INFO
>>>> Supported: replaces, timer Content-Type: application/sdp
>>>> Content-Length: 330 Path: <sip:[Loadbalancer];lr>
>>>>
>>>> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
>>>> s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio
>>>> 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3
>>>> GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
>>>> a=sendrecv a=nortpproxy:yes
>>>>
>>>> 09:49:09.258109 IP [Loadbalancer].5060 > [PRIVATE IP of
>>>> UAC].5060:
>>> SIP, length: 1307
>>>> E..7.. at .@.. at ..,+.........#>.INVITE sip:435572949012@[PRIVATE IP
>>>> of
>>> UAC];user=phone SIP/2.0
>>>> Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
>>>> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via:
>>>> SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via:
>>>> SIP/2.0/UDP
>> [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
>>
>>> d51.bf3830b5.0
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
>>> UAC]:64224">
>>>> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN
>>>>
>>> Gateway]>;tag=as33339e2e
>>>> To: <sip:+435572949012@[Proxy+Registrar]> Contact:
>>>> <sip:4366480588134@[PSTN Gateway]> Call-ID:
>>>> 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102
>>>> INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009
>>>> 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>>> SUBSCRIBE, NOTIFY,
>>> INFO
>>>> Supported: replaces, timer Content-Type: application/sdp
>>>> Content-Length: 330 Path: <sip:[Loadbalancer];lr>
>>>>
>>>> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
>>>> s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio
>>>> 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3
>>>> GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
>>>> a=sendrecv a=nortpproxy:yes
>>>>
>>>> 09:49:11.258107 IP [Loadbalancer].5060 > [PRIVATE IP of
>>>> UAC].5060:
>>> SIP, length: 1307
>>>> E..7.. at .@.. at ..,+.........#>.INVITE sip:435572949012@[PRIVATE IP
>>>> of
>>> UAC];user=phone SIP/2.0
>>>> Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes>
>>>> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via:
>>>> SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via:
>>>> SIP/2.0/UDP
>> [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3
>>
>>> d51.bf3830b5.0
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of
>>> UAC]:64224">
>>>> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN
>>>>
>>> Gateway]>;tag=as33339e2e
>>>> To: <sip:+435572949012@[Proxy+Registrar]> Contact:
>>>> <sip:4366480588134@[PSTN Gateway]> Call-ID:
>>>> 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102
>>>> INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009
>>>> 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>>>> SUBSCRIBE, NOTIFY,
>>> INFO
>>>> Supported: replaces, timer Content-Type: application/sdp
>>>> Content-Length: 330 Path: <sip:[Loadbalancer];lr>
>>>>
>>>> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway]
>>>> s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio
>>>> 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3
>>>> GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
>>>> a=sendrecv a=nortpproxy:yes
>>>>
>>>> 09:49:15.023388 IP [Proxy+Registrar].5060 >
>>>> [Loadbalancer].5060:
>> SIP,
>>> length: 397
>>>> E..... at .?.\"..,F..,+......a.CANCEL sip:435572949012@[PRIVATE IP
>>>> of
>>> UAC];user=phone SIP/2.0
>>>> Via: SIP/2.0/UDP
>>>> [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From:
>>>> "4366480588134" <sip:4366480588134@[PSTN
>>> Gateway]>;tag=as33339e2e
>>>> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To:
>>>> <sip:+435572949012@[Proxy+Registrar]> CSeq: 102 CANCEL
>>>> Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls
>>>> (x86_64/linux)) Content-Length: 0
>>>>
>>>>
>>>> 09:49:15.024138 IP [Loadbalancer].5060 >
>>>> [Proxy+Registrar].5060:
>> SIP,
>>> length: 413
>>>> E..... at .@.[...,+..,F........SIP/2.0 200 canceling Via:
>>>> SIP/2.0/UDP
>> [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
>>
>>> oxy+Registrar]
>>>> From: "4366480588134" <sip:4366480588134@[PSTN
>>> Gateway]>;tag=as33339e2e
>>>> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To:
>> <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
>>
>>> 25e-db87
>>>> CSeq: 102 CANCEL Server: Kamailio (1.5.1-notls (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> 09:49:15.024576 IP [Loadbalancer].5060 >
>>>> [Proxy+Registrar].5060:
>> SIP,
>>> length: 493
>>>> E.. .. at .@.Z...,+..,F........SIP/2.0 487 Request Terminated
>>>> Via: SIP/2.0/UDP
>> [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Pr
>>
>>> oxy+Registrar]
>>>> Via: SIP/2.0/UDP [PSTN
>>> Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060
>>>> From: "4366480588134" <sip:4366480588134@[PSTN
>>> Gateway]>;tag=as33339e2e
>>>> To:
>> <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
>>
>>> 25e-db87
>>>> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq:
>>>> 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> 09:49:15.025180 IP [Proxy+Registrar].5060 >
>>>> [Loadbalancer].5060:
>> SIP,
>>> length: 433
>>>> E..... at .?.[...,F..,+.......MACK sip:435572949012@[PRIVATE IP of
>>>>
>>> UAC];user=phone SIP/2.0
>>>> Via: SIP/2.0/UDP
>>>> [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From:
>>>> "4366480588134" <sip:4366480588134@[PSTN
>>> Gateway]>;tag=as33339e2e
>>>> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To:
>> <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec98
>>
>>> 25e-db87
>>>> CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls
>>>> (x86_64/linux)) Content-Length: 0
>>>>
>>>>
>>>> ----------------This is the config of my
>>>> loadbalancer--------------
>> --
>>> ----:
>>>> ####### Global Parameters #########
>>>>
>>>> debug=100 log_stderror=yes log_facility=LOG_LOCAL0
>>>>
>>>> fork=no children=4
>>>>
>>>> disable_tcp=yes
>>>>
>>>> port=5060
>>>>
>>>> listen=[Loadbalancer]
>>>>
>>>> ####### Modules Section ########
>>>>
>>>> mpath="/usr/lib/kamailio/modules/"
>>>>
>>>> loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so"
>>>> loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so"
>>>> loadmodule "textops.so" loadmodule "siputils.so" loadmodule
>>>> "xlog.so" loadmodule "nathelper.so" loadmodule "dispatcher.so"
>>>> loadmodule "path.so"
>>>>
>>>> # ----------------- setting module-specific parameters
>>>> ------------
>> --
>>> -
>>>>
>>>> # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name",
>>>> "/tmp/kamailio_fifo")
>>>>
>>>>
>>>> # ----- rr params ----- # add value to ;lr param to cope with
>>>> most of the UAs #modparam("rr", "enable_full_lr", 1) # do not
>>>> append from tag to the RR (no need for this script)
>>>> #modparam("rr", "append_fromtag", 0)
>>>>
>>>>
>>>> # -- nathelper modparam("nathelper", "rtpproxy_sock",
>>> "/var/run/rtpproxy/rtpproxy.sock")
>>>> modparam("nathelper", "received_avp", "$avp(i:80)")
>>>>
>>>> # -- dispatcher params -- modparam("dispatcher", "list_file",
>> "/etc/kamailio/dispatcher.list")
>>>> # -- path params -- modparam("path", "use_received", 1)
>>>>
>>>> ####### Routing Logic ########
>>>>
>>>>
>>>> # main request routing logic
>>>>
>>>> route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if
>>>> (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too
>>>> Many Hops"); exit; }
>>>>
>>>> # NAT detection route(4);
>>>>
>>>> if (has_totag()) { xlog("L_ALERT", "_____HASTOTAG [$rm] from
>>>> [$fu] to
>>> [$tu]\n");
>>>> # sequential request withing a dialog should # take the path
>>>> determined by record-routing if (loose_route()) {
>>>> xlog("L_ALERT", "_____LOOSE_ROUTE [$rm]
>> from
>>> [$fu] to [$tu]\n");
>>>> t_relay(); exit; } else { if ( is_method("ACK") ) { if (
>>>> t_check_trans() ) { xlog("L_ALERT", "_____NON-
>>> LOOSE, STATEFUL-ACK [$rm] from [$fu] to [$tu]\n");
>>>> # non loose-route, but
>>> stateful ACK; must be an ACK after a 487 or e.g. 404 from
>>> upstream server
>>>> t_relay(); exit; } else { xlog("L_ALERT",
>>> "_____ACK_WITHOUT_TRANSACTION [$rm] from [$fu] to [$tu]\n");
>>>> # ACK without matching
>>> transaction ... ignore and discard.\n");
>>>> exit; } } sl_send_reply("404","Not here"); } exit; }
>>>>
>>>> #initial requests
>>>>
>>>> # CANCEL processing if (is_method("CANCEL")) { if
>>>> (t_check_trans()) t_relay(); exit; }
>>>>
>>>> t_check_trans();
>>>>
>>>> # record routing if (!is_method("REGISTER|MESSAGE"))
>>>> xlog("L_ALERT", "_____RECORD_ROUTING [$rm] from
>> [$fu]
>>> to [$tu]\n");
>>>> record_route();
>>>>
>>>> #if ($rU==NULL) { # # request with no Username in RURI #
>>>> sl_send_reply("484","Address Incomplete"); # exit; #}
>>>>
>>>> if (check_route_param("nat=yes")) { setbflag(6); } #if
>>>> (isflagset(5) || isbflagset(6)) { route(5); #}
>>>>
>>>> /* example how to enable some additional event routes */ if
>>>> (is_method("INVITE")) { t_on_reply("1"); t_on_failure("1"); }
>>>>
>>>> if(!ds_is_from_list()) { ds_select_domain("0", "0"); } if
>>>> (!t_relay()) { sl_reply_error(); } }
>>>>
>>>>
>>>> # Caller NAT detection route route[4]{ force_rport(); if
>>>> (nat_uac_test("19")) { if (method=="REGISTER") {
>>>> add_path_received(); fix_nated_register(); } else {
>>>> fix_nated_contact(); } setflag(5); } else { add_path(); }
>>>> return; }
>>>>
>>>> # RTPProxy control route[5] { if (is_method("BYE")) {
>>>> unforce_rtp_proxy(); } else if (is_method("INVITE")){
>>>> force_rtp_proxy(); } if (!has_totag())
>>>> add_rr_param(";nat=yes"); return; }
>>>>
>>>> onreply_route[1] { xdbg("incoming reply\n");
>>>>
>>>> #if ((isflagset(5) || isbflagset(6)) &&
>> status=~"(183)|(2[0-
>>> 9][0-9])") {
>>>> if (status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if
>>>> (isbflagset(6)) { fix_nated_contact(); } }
>>>>
>>>> failure_route[1] { #if (is_method("INVITE") && (isbflagset(6)
>>>> ||
>> isflagset(5)))
>>> {
>>>> if (is_method("INVITE")) { unforce_rtp_proxy(); }
>>>>
>>>> if (t_was_cancelled()) { exit; } }
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________ Kamailio
>>>> (OpenSER) - Users mailing list Users at lists.kamailio.org
>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>>
>> _______________________________________________ Kamailio (OpenSER)
>> - Users mailing list Users at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________ Kamailio (OpenSER) -
> Users mailing list Users at lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
More information about the Users
mailing list