Hi Matt,
When a non-NATed incoming call to a NATed client is processed (INVITE), you
must make sure that you have a t_on_reply("1"); before you call t_relay (or
forward). The INVITE will not be detected as behind a NAT, but the
destination is (flag is set), and the reply will take care of the rewrite.
In your config, it looks like you call t_relay before setting
t_on_reply("1"); further down. A forward will only forward the SIP INVITE
to another SIP proxy for processing.
Paul (Java Rockx) just recently posted his config file with a working
NAThelper/RTPproxy setup. I suggest you look at the call logic found there.
His config is also easy to read with a lot of nice headers
I haven't tested RTP proxy between a client behind NAT and Asterisk, but
I believe that as long as you record-route the INVITE (as you do) and handle
the replies properly, it should work.
g-)
> Matt Schulte wrote:
>> Another note to this, I moved my 'forward' and lookup statements down
>> below the t_onreply statement. I figured this should allow ser to see
>> that the client is in fact behind a NAT. It catches that now however
>> I see this in my debug (ser):
>>
>> ser[21770]: transaction was sent to a NATED client -> fix nated
>> contact ser[21770]: ERROR: on_reply processing failed
>>
>> Could the last error be a/the problem? Come on I know someone else
>> has had this problem. Please help!
>> NOTE: I just tested this out on Asterisk (as a client behind NAT) and
>> got the same results. It's simply not changing the RTP IP address..
>>
>> --snippet--
>>
>> onreply_route[1] {
>> # NATed transaction ?
>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
>> log(1, "transaction was sent to a NATED client -> fix nated
>> contact\n");
>> fix_nated_contact();
>> force_rtp_proxy();
>> # otherwise, is it a transaction behind a NAT and we did not
>> # know at time of request processing ? (RFC1918 contacts)
>> } else if (nat_uac_test("1")) {
>> fix_nated_contact();
>> };
>>
>>
>> -----Original Message-----
>> From: Matt Schulte
>> Sent: Thursday, November 18, 2004 8:09 AM
>> To: serusers(a)lists.iptel.org
>> Subject: [Serusers] NATHelper + usrloc (+ rtpproxy?)
>>
>>
>> All,
>>
>> This is my first post to this list so go easy on me. :-) I'm
>> rather new to Ser, in fact I just installed it for the first time
>> early in the week. I'm working on the NAThelper module to get
>> traversal working, I have outbound (sip phone -> NATout -> ser)
>> working just peachy, RTP works in both directions hooray. The
>> question is I'm having problems getting RTP inbound, the ring of
>> course goes through, and RTP from the NAT'd side of course works fine
>> however getting back through the NAT (from outside) for RTP in this
>> sense fails. Let me explain the setup:
>>
>> I'm using the registrar, NAThelper, usrloc, and of course
>> (Portaone's) RTPproxy modules. The current SIP phone is an SNOM (yes
>> yes, I know..). The "endpoint" is Asterisk. When I do a sip debug on
>> Asterisk, I see the RTP request however it's coming from the NAT'd
>> fake address:
>>
>> v=0
>> o=root 780961119 780961119 IN IP4 192.168.1.101
>> s=call
>> c=IN IP4 192.168.1.101
>> t=0 0
>> m=audio 10004 RTP/AVP 0
>> a=rtpmap:0 pcmu/8000
>> a=sendrecv
>>
>>
>> I have an idea of what to fix just not sure how to fix it. Obviously
>> we need it to goto RTPproxy, since this is "backwards" how would I
>> get it to recognize the correct IP?
>>
>> See my config below, most of it is ripped off of the NAThelper.cfg
>> example. :-) Thanks all..
>>
>> NOTE: All calls are destined for ${SIPDOMAIN}, in this case, the
>> machines hostname. This is normal and intentional :-)
>>
>> # ---- SNIPPAGE ----
>> modparam("rr", "enable_full_lr", 1)
>>
>> # !! Nathelper
>> modparam("registrar", "nat_flag", 6)
>> modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
>> modparam("nathelper", "ping_nated_only", 1) # Ping only clients
>> behind NAT
>> # main routing logic
>>
>> route{
>>
>> # initial sanity checks -- messages with
>> # max_forwards==0, or excessively long requests
>> if (!mf_process_maxfwd_header("10")) {
>> sl_send_reply("483","Too Many Hops");
>> break;
>> };
>> if (msg:len >= max_len ) {
>> sl_send_reply("513", "Message too big");
>> break;
>> };
>> # !! Nathelper
>> # Special handling for NATed clients; first, NAT test is
>> # executed: it looks for via!=received and RFC1918 addresses
>> # in Contact (may fail if line-folding is used); also,
>> # the received test should, if completed, should check all
>> # vias for rpesence of received
>> if (nat_uac_test("3")) {
>> # Allow RR-ed requests, as these may indicate that
>> # a NAT-enabled proxy takes care of it; unless it is
>> # a REGISTER
>> log("LOG: Caught uac test 3 \n");
>> if (method == "REGISTER" || !
>> search("^Record-Route:")) {
>> log("LOG: Someone trying to register from private
>> IP, rewriting\n");
>>
>> # This will work only for user agents that
>> support symmetric
>> # communication. We tested quite many of them and
>> majority is
>> # smart enough to be symmetric. In some phones it
>> takes a configuration
>> # option. With Cisco 7960, it is called
>> NAT_Enable=Yes, with kphone it is
>> # called "symmetric media" and "symmetric
>> signalling".
>>
>> fix_nated_contact(); # Rewrite contact with
>> source IP of signalling
>> if (method == "INVITE") {
>> log("LOG: fix nated sdp\n");
>> fix_nated_sdp("1"); # Add direction=active to
>> SDP
>> };
>> force_rport(); # Add rport parameter to topmost
>> Via setflag(6); # Mark as NATed
>> };
>> };
>>
>> # we record-route all messages -- to make sure that
>> # subsequent messages will go through our proxy; that's
>> # particularly good if upstream and downstream entities
>> # use different transport protocol
>> if (!method=="REGISTER") record_route();
>>
>> # subsequent messages withing a dialog should take the
>> # path determined by record-routing
>> if (loose_route()) {
>> # mark routing logic in request
>> append_hf("P-hint: rr-enforced\r\n");
>> route(1);
>> break;
>> };
>>
>> if (!uri==myself) {
>> # mark routing logic in request
>> append_hf("P-hint: outbound\r\n");
>> route(1);
>> break;
>> };
>>
>>
>> if (uri==myself) {
>>
>> if (method=="REGISTER") {
>> log("LOG: Caught register, registering user
>> in local db\n");
>> save("location");
>> break;
>> };
>>
>> lookup("aliases");
>> if (!uri==myself) {
>> append_hf("P-hint: outbound alias\r\n");
>> route(1);
>> break;
>> };
>> log("LOG: Caught uri myself\n");
>> # native SIP destinations are handled using our
>> USRLOC DB
>> #if (!lookup("location")) {
>> # sl_send_reply("404", "Do what now");
>> # break;
>> #};
>> };
>> append_hf("P-hint: usrloc applied\r\n");
>> route(1);
>>
>> }
>>
>> route[1]
>> {
>> # !! Nathelper
>> if
>> (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
>> !search("^Route:")){ sl_send_reply("479", "We don't
>> forward to private IP addresses");
>> break;
>> };
>> # if client or server know to be behind a NAT, enable relay
>> if (isflagset(6)) {
>> log("LOG: Caught NAT flag 6 forcing rtp proxy\n");
>> force_rtp_proxy();
>> };
>> if (method=="REGISTER") {
>> break;
>> log("LOG: Caught Register down in our call routing,
>> breaking\n");
>> };
>>
>> #### Below is mostly my own doing ####
>> if (method=="INVITE") {
>> log("LOG: Caught INVITE \n");
>> if (lookup("location")) {
>> log ("LOG: Caught registered invite, sending
>> there\n");
>> # NOTE forcing rtp maybe bad idea for ALL
>> users, this is
>> # a quick fix (which doesn't work anyway!)
>> #force_rtp_proxy();
>> #forward(uri:host, uri:port); #nor does this
>> t_relay();
>> break;
>> } else if (uri=~"^sip:[0-9]*@") { # ... forward to
>> asterisk;
>> forward(xxx.xxx.xxx.xxx, 5060);
>> log("LOG: Tapping rowlf\n");
>> break;
>> };
>> };
>> #### ####
>>
>> t_on_reply("1");
>>
>> if (!t_relay()) {
>> sl_reply_error();
>> };
>> }
>>
>> # !! Nathelper
>> onreply_route[1] {
>> # NATed transaction ?
>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
>> fix_nated_contact();
>> force_rtp_proxy();
>> # otherwise, is it a transaction behind a NAT and we did not
>> # know at time of request processing ? (RFC1918 contacts)
>> } else if (nat_uac_test("1")) {
>> fix_nated_contact();
>> };
>> }
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi everybody,
I tried logining in using username:admin passwd:heslo but I am unable to
login to the system.I have created users using serctl and tried but it
is not working can anybody please let me know how to solve this serweb
login problem........................
Regrads,
nihar.
_______________________________________________
Join Excite! - http://www.excite.com
The most personalized portal on the Web!
Hi everybody,
I am unable to login to the serweb console when I check the httpd logfiles it gives the following errors and can any one tell me what is
username and passwd to login to serweb I tried with user:ser passwd:heslo but it is not logining in and not giving any errors on the
screen is there any way to create users please let me know.
Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/prolog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/separator.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/epilog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/prolog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/separator.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/epilog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
Regards,
nihar.
_______________________________________________
Join Excite! - http://www.excite.com
The most personalized portal on the Web!
Hello,
we have (again) problems with a Cisco 7960 SIP phone.
#
U 2004/11/17 13:18:55.664760 1.2.3.4:50520 -> 11.22.33.44:5060
INVITE sip:07216636445@testbed.de SIP/2.0.
Via: SIP/2.0/UDP 1.2.3.4:5065;branch=z9hG4bK218ce466.
From: "Koenig"
<sip:koenig@testbed.de>;tag=0006283e0a68009a4d72450f-0d8cfd54.
To: <sip:07216636445@testbed.de>.
Call-ID: 0006283e-0a680093-034392eb-37bfeedf(a)1.2.3.4.
CSeq: 101 INVITE.
User-Agent: CSCO/7.
Contact: <sip:koenig@1.2.3.4:5065>.
Expires: 180.
Content-Type: application/sdp.
Content-Length: 246.
Accept: application/sdp.
Remote-Party-ID: "Koenig"
<sip:koenig@1.2.3.4>;party=calling;id-type=subscriber;privacy=off;screen=no.
.
(sdp stripped)
#
U 2004/11/17 13:18:55.665068 11.22.33.44:5060 -> 1.2.3.4:5065
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 1.2.3.4:5065;branch=z9hG4bK218ce466.
From: "Koenig"
<sip:koenig@testbed.de>;tag=0006283e0a68009a4d72450f-0d8cfd54.
To: <sip:07216636445@testbed.de>;tag=16ac3fc2258766c821c391b58b08db64.371f.
Call-ID: 0006283e-0a680093-034392eb-37bfeedf(a)1.2.3.4.
CSeq: 101 INVITE.
Proxy-Authenticate: Digest realm="testbed.de",
nonce="419b42db55c1a65dac6b825b8c2f8bfa62539beb", qop="auth".
Server: Sip EXpress router (0.8.14-2 (i386/linux)).
Content-Length: 0.
Warning: 392 11.22.33.44:5060 "Noisy feedback tells: pid=2660
req_src_ip=1.2.3.4 req_src_port=50520 in_uri=sip:07216636445@testbed.de
out_uri=sip:07216636445@testbed.de via_cnt==1".
.
#
U 2004/11/17 13:18:55.764578 1.2.3.4:51368 -> 11.22.33.44:5060
ACK sip:07216636445@testbed.de SIP/2.0.
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK218ce466.
From: "Koenig"
<sip:koenig@testbed.de>;tag=0006283e0a68009a4d72450f-0d8cfd54.
To: <sip:07216636445@testbed.de>;tag=16ac3fc2258766c821c391b58b08db64.371f.
Call-ID: 0006283e-0a680093-034392eb-37bfeedf(a)1.2.3.4.
CSeq: 101 ACK.
Content-Length: 0.
.
#
U 2004/11/17 13:18:55.764915 11.22.33.44:5060 -> 11.22.33.55:5060
ACK sip:+497216636445@213.218.10.130:5060 SIP/2.0.
Max-Forwards: 10.
Via: SIP/2.0/UDP 11.22.33.44;branch=0.
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK218ce466.
From: "Koenig"
<sip:koenig@testbed.de>;tag=0006283e0a68009a4d72450f-0d8cfd54.
To: <sip:07216636445@testbed.de>;tag=16ac3fc2258766c821c391b58b08db64.371f.
Call-ID: 0006283e-0a680093-034392eb-37bfeedf(a)1.2.3.4.
CSeq: 101 ACK.
Content-Length: 0.
.
As you can see, the ACK for the 407 gets forwarded by SER. It does not match
the previous transaction. This should not be the case. Is there anything we
can do about this, except flying to munich and beat on cisco?
Best regards,
Martin
Hi ahmed,
I tried logining using username:admin passwd:heslo but it still unable
to login into the serweb can you tell what is the problem and can you
tell me where can I find epilog.html,seperator.html,prolog.html because
in httpd log it is showing this error files and does not come with the
package
1)webserver is running
2) mysql server is runing
3)application is deployed on the webserver
Regards,
nihar.
_______________________________________________
Join Excite! - http://www.excite.com
The most personalized portal on the Web!
Hi everybody,
I am unable to login to the serweb console when I check the httpd logfiles it gives the following errors and can any one tell me what is
username and passwd to login to serweb I tried with user:ser passwd:heslo but it is not logining in and not giving any errors on the
screen is there any way to create users please let me know.
Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/prolog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/separator.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/epilog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:29 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/prolog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/separator.html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/iptel/html, referer: http://localhost/iptel/serweb/html/admin/index.php
[Fri Nov 19 14:07:46 2004] [error] [client 127.0.0.1] File does not exist: /var/www/html/epilog.html, referer: http://localhost/iptel/serweb/html/admin/index.php
Regards,
nihar.
_______________________________________________
Join Excite! - http://www.excite.com
The most personalized portal on the Web!
Another note to this, I moved my 'forward' and lookup statements down
below the t_onreply statement. I figured this should allow ser to see
that the client is in fact behind a NAT. It catches that now however I
see this in my debug (ser):
ser[21770]: transaction was sent to a NATED client -> fix nated contact
ser[21770]: ERROR: on_reply processing failed
Could the last error be a/the problem? Come on I know someone else has
had this problem. Please help!
NOTE: I just tested this out on Asterisk (as a client behind NAT) and
got the same results. It's simply not changing the RTP IP address..
--snippet--
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
log(1, "transaction was sent to a NATED client -> fix nated
contact\n");
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
-----Original Message-----
From: Matt Schulte
Sent: Thursday, November 18, 2004 8:09 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] NATHelper + usrloc (+ rtpproxy?)
All,
This is my first post to this list so go easy on me. :-) I'm
rather new to Ser, in fact I just installed it for the first time early
in the week. I'm working on the NAThelper module to get traversal
working, I have outbound (sip phone -> NATout -> ser) working just
peachy, RTP works in both directions hooray. The question is I'm having
problems getting RTP inbound, the ring of course goes through, and RTP
from the NAT'd side of course works fine however getting back through
the NAT (from outside) for RTP in this sense fails. Let me explain the
setup:
I'm using the registrar, NAThelper, usrloc, and of course (Portaone's)
RTPproxy modules. The current SIP phone is an SNOM (yes yes, I know..).
The "endpoint" is Asterisk. When I do a sip debug on Asterisk, I see the
RTP request however it's coming from the NAT'd fake address:
v=0
o=root 780961119 780961119 IN IP4 192.168.1.101
s=call
c=IN IP4 192.168.1.101
t=0 0
m=audio 10004 RTP/AVP 0
a=rtpmap:0 pcmu/8000
a=sendrecv
I have an idea of what to fix just not sure how to fix it. Obviously we
need it to goto RTPproxy, since this is "backwards" how would I get it
to recognize the correct IP?
See my config below, most of it is ripped off of the NAThelper.cfg
example. :-) Thanks all..
NOTE: All calls are destined for ${SIPDOMAIN}, in this case, the
machines hostname. This is normal and intentional :-)
# ---- SNIPPAGE ----
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind
NAT
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
log("LOG: Caught uac test 3 \n");
if (method == "REGISTER" || ! search("^Record-Route:"))
{
log("LOG: Someone trying to register from private
IP, rewriting\n");
# This will work only for user agents that support
symmetric
# communication. We tested quite many of them and
majority is
# smart enough to be symmetric. In some phones it
takes a configuration
# option. With Cisco 7960, it is called
NAT_Enable=Yes, with kphone it is
# called "symmetric media" and "symmetric
signalling".
fix_nated_contact(); # Rewrite contact with source
IP of signalling
if (method == "INVITE") {
log("LOG: fix nated sdp\n");
fix_nated_sdp("1"); # Add direction=active to
SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
if (uri==myself) {
if (method=="REGISTER") {
log("LOG: Caught register, registering user in
local db\n");
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
log("LOG: Caught uri myself\n");
# native SIP destinations are handled using our USRLOC
DB
#if (!lookup("location")) {
# sl_send_reply("404", "Do what now");
# break;
#};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&& !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
log("LOG: Caught NAT flag 6 forcing rtp proxy\n");
force_rtp_proxy();
};
if (method=="REGISTER") {
break;
log("LOG: Caught Register down in our call routing,
breaking\n");
};
#### Below is mostly my own doing ####
if (method=="INVITE") {
log("LOG: Caught INVITE \n");
if (lookup("location")) {
log ("LOG: Caught registered invite, sending
there\n");
# NOTE forcing rtp maybe bad idea for ALL users,
this is
# a quick fix (which doesn't work anyway!)
#force_rtp_proxy();
#forward(uri:host, uri:port); #nor does this
t_relay();
break;
} else if (uri=~"^sip:[0-9]*@") { # ... forward to
asterisk;
forward(xxx.xxx.xxx.xxx, 5060);
log("LOG: Tapping rowlf\n");
break;
};
};
#### ####
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Given that the asterisk guys often don't seem to care too much for SER I
think I will probably have better luck getting this question addressed
here. Can anyone provide a brief explanation of the issue I raise below?
The goal is to build least cost routing, accounting, etc in Asterisk. I
have seen this referred to as using asterisk like a b2bua in conversations
on this list and in other documentation.
--
Tracy Reed http://copilotcom.com
This message is cryptographically signed for your protection.
Info: http://copilotconsulting.com/sig
Hi all.
I see this message in my syslog
Nov 17 10:50:00 sip01 /usr/local/sbin/ser[30968]: WARNING:fm_free: free(0) called
Is this a something that I should be concerned with?
Regards,
Paul
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