What is the best way to detect whether an INVITE message came from a certain
IP address?
I.e. if a call comes from a certain asterisk box?
I have thought of using the textops search(), but this seems a little
clumsy.
Perhaps there is a way to get directly at the via headers?
Or is there a keyword of corresponding functionality to the uri keyword?
Thanks,
Mark Price
hi all
i have problem in configurations of mediaproxy module i have download mediaproxy in my ser server and i have make one file name(mediaproxy.ini) in mediaproxy dir and do some change and run this command
mediaproxy start
nothing can heppand please help me out on this because at this time i am in very bad conditions
thanks
vijay
---------------------------------
Vijay Tiwari
Network Admin
Sachitel Communications
Noida(U.P.)
M- 09871633711
---------------------------------
Everyone is raving about the all-new Yahoo! Mail.
I just posted about a problem with rtpproxy, but i was able to correct
that. now ser is able to connect to rtpproxy, but media is still
being lost.
someone suggested i put my external ip in the listen= value at the
top, but that breaks remote registration.
I am following the rtpproxy section of the "Getting Started with Ser
Docs" http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html so
one would expect i would have ser.cfg set up correctly, but obviously
something is wrong.
here is some ngrep output. 10.0.0.10 is SER. 10.0.0.110 is a client
LOCAL to ser, and 10.0.0.100/67.84.215.54 is a remote client behind
yet another nat
U 10.0.0.10:6060 -> 10.0.0.110:9267
INVITE sip:1000@10.0.0.110:9267 SIP/2.0.
Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>.
To: <sip:1000@bmy.gotdns.org>.
From: <sip:1001@bmy.gotdns.org>;tag=d752304f.
Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1.
Via: SIP/2.0/UDP
10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170.
Call-ID: cc766a6d3e104232.
CSeq: 2 INVITE.
Contact: <sip:1001@67.84.215.54:9170>.
Max-Forwards: 16.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
User-Agent: eyeBeam release 3004w stamp 16863.
Content-Length: 285.
.
v=0.
o=- 105355536 105355589 IN IP4 10.0.0.100.
s=eyeBeam.
c=IN IP4 10.0.0.10.
t=0 0.
m=audio 20026 RTP/AVP 100 6 0 8 3 18 5 101.
a=alt:1 1 : 7D3DDAD1 578F25CA 10.0.0.100 9172.
a=fmtp:101 0-15.
a=rtpmap:100 speex/16000.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
a=nortpproxy:yes.
#
U 10.0.0.110:9267 -> 10.0.0.10:6060
SIP/2.0 180 Ringing.
To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b.
From: <sip:1001@bmy.gotdns.org>;tag=d752304f.
Via: SIP/2.0/UDP
10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10.
Via: SIP/2.0/UDP
10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170.
Call-ID: cc766a6d3e104232.
CSeq: 2 INVITE.
Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>.
Contact: <sip:1000@10.0.0.110:9267>.
Content-Length: 0.
.
#
U 10.0.0.10:6060 -> 67.84.215.54:9170
SIP/2.0 180 Ringing.
To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b.
From: <sip:1001@bmy.gotdns.org>;tag=d752304f.
Via: SIP/2.0/UDP
10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170.
Call-ID: cc766a6d3e104232.
CSeq: 2 INVITE.
Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>.
Contact: <sip:1000@10.0.0.110:9267>.
Content-Length: 0.
.
#
U 10.0.0.110:9267 -> 10.0.0.10:6060
SIP/2.0 180 Ringing.
To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b.
From: <sip:1001@bmy.gotdns.org>;tag=d752304f.
Via: SIP/2.0/UDP
10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10.
Via: SIP/2.0/UDP
10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170.
Call-ID: cc766a6d3e104232.
CSeq: 2 INVITE.
Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>.
Contact: <sip:1000@10.0.0.110:9267>.
Content-Length: 0.
.
#
U 10.0.0.10:6060 -> 67.84.215.54:9170
SIP/2.0 180 Ringing.
To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b.
From: <sip:1001@bmy.gotdns.org>;tag=d752304f.
Via: SIP/2.0/UDP
10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170.
Call-ID: cc766a6d3e104232.
CSeq: 2 INVITE.
Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>.
Contact: <sip:1000@10.0.0.110:9267>.
Content-Length: 0.
and here is my ser.cfg which ive also attached.
#
# $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=10.0.0.10
alias=hostname.dyndns.org
port=6060
children=4
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://user:pw@localhost/ser"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
#From Getting Started
modparam("auth_db|uri_db|usrloc", "db_url", "mysql://user:pw@localhost/ser")
# ----------------- setting module-specific parameters ---------------
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
modparam("nathelper", "rtpproxy_disable", 1)
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
modparam("usrloc", "db_mode", 2)
modparam("registrar", "nat_flag", 6)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# 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 >= 2048 ) {
sl_send_reply("513", "Message too big");
break;
};
# 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();
};
if (method=="BYE" || method=="CANCEL") {
unforce_rtp_proxy();
}
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
if ((method=="INVITE" || method=="REFER") && !has_totag()) {
sl_send_reply("403", "Forbidden");
break;
};
if (method=="INVITE") {
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
}else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
if (nat_uac_test("19")) {
setflag(6);
force_rport();
fix_nated_contact();
};
force_rtp_proxy("l");
};
route(1);
break;
};
#======= call type processing section
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(4);
route(1);
break;
};
if (method=="ACK") {
route(1);
break;
} else if (method=="CANCEL") {
route(1);
break;
} else if (method=="INVITE") {
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(1);
}
route[1]
{
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
if (method=="INVITE" || isflagset(6)) {
unforce_rtp_proxy();
};
sl_reply_error();
};
}
route[2] {
#-------------------------
#REGISTER Message Handler
#-------------------------
if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) {
setflag(6);
fix_nated_register();
force_rport();
};
sl_send_reply("100", "Trying");
if (!www_authorize("","subscriber")) {
www_challenge("","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
#----------
#--INVITE Message Handler
#-----------
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
if (nat_uac_test("19")) {
setflag(6);
}
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(4);
route(1);
}
route[4] {
#NAT Traversal
if (isflagset(6)) {
force_rport();
fix_nated_contact();
force_rtp_proxy();
}
}
onreply_route[1] {
if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
if (!search("^Contact-Length:[ ]*0")) {
force_rtp_proxy();
};
};
if (nat_uac_test("1")) {
fix_nated_contact();
};
}
Hi,
I was wondering about the correct handeling of an 484 address incomplete
message.
I've seen different handeling from UAC to UAC
Problem:
My grandstreams receives a 484 and returns an ACK. This is how it
should.And i hear beep-beep tones.
the moment I terminate (on hook), the UAC sends a CANCEL message which
generates a baaaad loop in my system....
1. Is this cancel according to RFC?
2. Is there a way to consume/drop such an cancel?
br hw
--
Helge Waastad
Senior Engineer
Systemavdelingen
Smartnet
Can someone please explain what the line
if ((@msg.supported=~"eventlist"))
is testing in the presence example code? This lines always returns false for
me and seemingly the XCAP server is not being queried then.
Thanks
_________________________________________________________________
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a chance to WIN $10,000! www.mobilejamfest.com
Hi All
I made an entry "10.25.119.156, any, ^sip:.*$" into trusted table so
that any packet sent from 10.25.119.156 would be trusted by SER and
don't ask for its credentials. But SER is replying 407- proxy auth
needed!! After this I tried with permissions.allow file:
"ALL : "sip:mann@imsfw.de$"" but still no progress same response from
SER.
My ser.cfg entries:
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
modparam("permissions", "source_col", "src_ip")
modparam("permissions", "proto_col", "proto")
modparam("permissions", "from_col", "from_pattern")
modparam("permissions",
"default_deny_file","/usr/lib/ser/modules/permissions.deny")
modparam("permissions",
"default_allow_file","/usr/lib/ser/modules/permissions.allow")
Is it correct? Please give your suggestions for the same.
Thanks in anticipation
Kamal Mann
Im following "Getting Started with SER" and I'm stuck on the chapter
about nat traversal
Ive tried both methods, currently im working on rtpptoxy. the exact
errors I get when i try to make calls from behind NAT are:
0(1731) WARNING: rtpp_test: can't get version of the RTP proxy
0(1731) WARNING: rtpp_test: support for RTP proxy has been disabled
0(1731) ERROR: force_rtp_proxy2: support for RTP proxy is disabled
i have compiled and installed rtpproxy, and made sure the permissions were set.
bmy bin # ps ax|grep rtp
1714 ? Ss 0:00 ./rtpproxy
bmy bin # ls -la /var/run/rtpproxy.*
-rw-r--r-- 1 root root 5 Nov 1 19:27 /var/run/rtpproxy.pid
srwxr-xr-x 1 root root 0 Nov 1 19:27 /var/run/rtpproxy.sock
i have attached my ser.cfg but below are some of the parts of it
debug=3 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=10.0.0.10
alias=hostname.dyndns.org
port=6060
children=4
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://user:pw@localhost/ser"
<skip>
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
<skip>
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 >= 2048 ) {
sl_send_reply("513", "Message too big");
break;
};
# 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();
};
if (method=="BYE" || method=="CANCEL") {
unforce_rtp_proxy();
}
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
if ((method=="INVITE" || method=="REFER") && !has_totag()) {
sl_send_reply("403", "Forbidden");
break;
};
if (method=="INVITE") {
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
}else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
if (nat_uac_test("19")) {
setflag(6);
force_rport();
fix_nated_contact();
};
force_rtp_proxy("l");
};
route(1);
break;
};
#======= call type processing section
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(4);
route(1);
break;
};
if (method=="ACK") {
route(1);
break;
} else if (method=="CANCEL") {
route(1);
break;
} else if (method=="INVITE") {
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(1);
}
route[1]
{
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
if (method=="INVITE" || isflagset(6)) {
unforce_rtp_proxy();
};
sl_reply_error();
};
}
route[2] {
#-------------------------
#REGISTER Message Handler
#-------------------------
if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) {
setflag(6);
fix_nated_register();
force_rport();
};
sl_send_reply("100", "Trying");
if (!www_authorize("","subscriber")) {
www_challenge("","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
#----------
#--INVITE Message Handler
#-----------
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
if (nat_uac_test("19")) {
setflag(6);
}
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(4);
route(1);
}
route[4] {
#NAT Traversal
if (isflagset(6)) {
force_rport();
fix_nated_contact();
force_rtp_proxy();
}
}
onreply_route[1] {
if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
if (!search("^Contact-Length:[ ]*0")) {
force_rtp_proxy();
};
};
if (nat_uac_test("1")) {
fix_nated_contact();
};
}
Hi all
I have installed CDROUTER working fine for 30days now
almost all iam able to see 120 users calling at a time no problem
but past 7days iam getting lot of voice breaking problems
and other side not able to hear my side voice
my Server config
Xeon with 1 GB Ram 86GB hdd
and most of the time looks like below
~# free
total used free shared buffers cached
Mem: 903484 897328 6156 0 1892 830208
-/+ buffers/cache: 65228 838256
Swap: 0 0 0
when i check top
Cpu use is 2% only
when i configure directly with provider using PAP2 , voice is clear
can some one guide me what is the problem
how can i test and how can i resolve this problem
help will be appriciate
Ram
Note: forwarded message attached.
Vijay Tiwari
Network Admin
Sachitel Communications
Noida(U.P.)
M- 09871633711
---------------------------------
Check out the New Yahoo! Mail - Fire up a more powerful email and get things done faster.
hi all
problem my mail are not coming on mail list
Vijay Tiwari
Network Admin
Sachitel Communications
Noida(U.P.)
M- 09871633711
---------------------------------
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