Sorry if I'm asking stupid question but I'm trying to set up alias for
example 1234<->john.doe and make it permanent. As far as I understand if
I use serctl add alias command and then contact for the person will
change (dynamic ip, login from other coputer etc) my alias would not
work anymore. Please tell me if I'm worng and if there is a way to fix
this problem.
Thank you,
Igor
I installed ser 8.14 and using mediaproxy 1.4.2.
I am trying to use X-lite to make a call to the PSTN through a PSTN
gateway (asterisk). The PSTN phone rings and get voice, but X-lite
only pass voice. I can hear nothing.
When I use the tool sessions.py available with mediaproxy, it shows me
the call but shows that it is inactive, although the PSTN phone can
hear audio. And after 60 seconds it drops the session, but the PSTN
end still can hear the audio sent from X-lite. And sessions.py does
not show the IP of asterisk....
[root@someplace mediaproxy]# ./sessions.py
Caller Via Called Status Duration
Codec Type Traffic
----------------------------------------------------------------------------------------------
200.150.x.x:8000 - 127.0.0.1:35008 - ?.?.?.?:? inactive 0'10"
Unknown Audio 0/0/0
If I run the rtpgenerator.py it shows the test rtp with traffic flowing.
Here it is what I do in ser.cfg:
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (!lookup("location")) {
if ( ( uri=~"^sip:[0-9]{8}@.*" ) ) {
t_relay_to_udp("X.X.X.X", "5060");
break;
}
sl_send_reply("404", "User not found");
break;
};
If I turn off mediaproxy and disable the use_media_proxy, everything works fine.
Any ideas? I read every possible document and found nothing about this.
Thanks
Felipe
--
Master Student - Electrical Engineering Department
Computer Engineering and Telecommunications Research Group
Universidade Federal de Minas Gerais - Brazil
"For God so loved the world that he gave his one and only Son, that
whoever believes in him shall not perish but have eternal life."
John 3:16
Here, try this script. I couldn't test it, but apart from minor errors it should work.
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
# must be yes since REGISTER processing causes an INVITE to be sent,
# which needs to be processed by another process
fork=yes
children=4
alias=server.sam.cse
alias=sam.cse
# debugging
log_stderror=yes # (cmd line: -E)
#mhomed=yes
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://root:vlftmd@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/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/exec.so"
# ----------------- setting module-specific parameters ---------------
modparam("rr", "enable_full_lr", 1)
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser")
# ------------------------- 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")) {
log(1, "ERROR: Too Many Hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
#keep this low to prevent possible DoS attacks and misconfigurations from overloading the server
if (msg:len>=2048) {
log(1, "ERROR: Message Too Big\n");
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 (loose_route()) {
route(1);
break;
};
if (method=="REGISTER")
{
log(1, "INFO: User Registered\n");
# save location before initiating welcome
save("location");
record_route();
sl_send_reply("200", "ok");
# welcome message
log(1, "INFO: Initiating Welcome Message \n");
route(3);
break;
};
log(1, "record-route processing completed\n");
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself)
{
if(method=="INVITE")
{
log(1, "INFO: SIP Invite");
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
log(1, "ERROR: User Not Found\n");
break;
};
};
record_route();
};
route(1);
}
route[1]
{
#send the call outward
if (!t_relay())
{
xlog("L_WARN", "ERROR: t_relay failed");
sl_reply_error();
};
}
# welcome message -- if a REGISTER succeeded, look if it is the first-time;
# if so, initiate click-to-dial transaction to set up call to an
announcement
# server; edit the config values first to make it work
route[3] {
if (!exec_msg('
# config:
# --announcement server URI
ANS="sip:mijin@sam.cse"
# --SIP domain
# DOMAIN="192.168.178.193"
DOMAIN="192.168.178.194"
# ctd
CTD=${HOME}/sip_router/examples/ctd.sh
# ------------------------------------
# check if first time ...
SIP_UID=`echo $SIP_HF_TO | sed -e
"s/^.*sip:\([a-zA-Z0-9_\.]*\)@.*$/\1/g"`
QUERY="select flag from subscriber
where username=\"$SIP_UID\";
update subscriber set flag=\"x\"
where username=\"$SIP_UID\" ";
mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null
# ... if so, c-t-d to announcement server
if [ "$?" -ne 0 ] ; then
# flag was not set to x yet -- first-time registration;
# initiate a call from telephone of the user to an announcement server
$CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1
fi
')) {
log(1, "ERROR: Welcome Message Script Failed\n");
}
log(1, "INFO: Welcome Message Sent\n");
}
#----------------End Script---------------------
---Mike
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org on behalf of ? ??
Sent: Fri 9/23/2005 6:30 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] SIP/2.0 483 Too Many Hops
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear....
I inserted command like this.
What I want is that
when I loaded test.cfg file, what I check SIP messages using sipp client
mode
I want 200 OK. But, as you see, I received "SIP/2.0 483 Too Many Hops"
error message.
So I was advanced about this.
- alias="sam.cse"//SIP_ DOMAIN
alias="server.sam.cse"//my_computer.SIP_DOMAIN
alias="192.168.178.194"//my_ip
- add my hostname in /etc/host
- change to : if (!mf_process_maxfwd_header("100")
but, I couldn't solve this problem.
I would really appreciate for giving some advise.
Thank you in advance.............................
mi-jin Song
=====================================================
[root@cse4 ~]# ser -f test.cfg
Listening on
udp: 127.0.0.1 [127.0.0.1]:5060
udp: 192.168.178.194 [192.168.178.194]:5060
tcp: 127.0.0.1 [127.0.0.1]:5060
tcp: 192.168.178.194 [192.168.178.194]:5060
Aliases:
tcp: server:5060
tcp: localhost:5060
tcp: localhost.localdomain:5060
tcp: cse4:5060
udp: server:5060
udp: localhost:5060
udp: localhost.localdomain:5060
udp: cse4:5060
*: 192.168.178.194:*
*: sam.cse:*
*: server.sam.cse:*
stateless - initializing
0(0) Maxfwd module- initializing
exec - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) ERROR: tcp_init: bind(6, 0x810dafc, 16) on 127.0.0.1: Address already
in use
=====================================================
[root@cse4 sipp]# ./sipp -sn uac 192.168.178.194
Resolving 192.168.178.194...
------------------------------ Scenario Screen -------- [1-4]: Change
Screen --
Call-rate(length) Port Total-time Total-calls Remote-host
10.0(0 ms)/1.000s 5061 2.54 s 25
192.168.178.194:5060(UDP)
5 new calls during 0.538 s period 2 ms scheduler resolution
0 concurrent calls (limit 30) Peak was 1 calls, after 0 s
0 out-of-call msg (discarded)
1 open sockets
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 25 0 0
100 <---------- 25 0 25
180 <---------- 0 0 0
200 <---------- E-RTD 0 0 0
ACK ----------> 0 0
[ 0 ms]
BYE ----------> 0 0 0
200 <---------- 0 0 0
------------------------------ Test Terminated
---------------------------------
----------------------------- Statistics Screen ------- [1-4]: Change
Screen --
Start Time | 2005-09-23 20:03:13
Last Reset Time | 2005-09-23 20:03:15
Current Time | 2005-09-23 20:03:16
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:536 | 00:00:02:543
Call Rate | 9.328 cps | 9.831 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 5 | 25
Total Call created | | 25
Current Call | 0 |
-------------------------+---------------------------+--------------------------
Successful call | 0 | 0
Failed call | 6 | 25
-------------------------+---------------------------+--------------------------
Response Time | 00:00:00:000 | 00:00:00:000
Call Length | 00:00:00:004 | 00:00:00:004
------------------------------ Test Terminated
--------------------------------
2005-09-23 20:03:16: Unexpected message for Call-ID
'25.4743.127.0.0.1(a)sipp.call.id': while expecting '100' response, received
'SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP 127.0.0.1:5061;received=192.168.178.194
From: sipp <sip:sipp@127.0.0.1:5061>;tag=25
To: sut
<sip:service@192.168.178.194:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6e68
Call-ID: 25.4743.127.0.0.1(a)sipp.call.id
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=4688
req_src_ip=192.168.178.194 req_src_port=5060
in_uri=sip:service@192.168.178.194:5060
out_uri=sip:service@192.168.178.194:5060 via_cnt==18"
' .
sipp: There were more errors, enable -trace_err to log them.
=======================================================
<< test.cfg >>
#
# $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $
#
# welcome message for new subscribers; based on exec
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
# must be yes since REGISTER processing causes an INVITE to be sent,
# which needs to be processed by another process
fork=yes
children=4
alias="server.sam.cse"
alias="sam.cse"
alias="192.168.178.194"
# debugging
log_stderror=yes # (cmd line: -E)
mhomed=yes
fifo="/tmp/ser_fifo"
# ------------------ 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/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/exec.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" )
modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser")
# ------------------------- request routing logic -------------------
# main routing logic
route{
log(1, "\n\n");
log(1, "START!!!\n");
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("15")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len>=204822) {
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=="INVITE") record_route(); # 1=loose routing
if(method=="INVITE"){
log(1, "receive INVITE\n");
record_route();
sl_send_reply("200", "ok");
};
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if(method=="ACK"){
# log(1, "receive ACK\n");
# };
log(1, "record-route processing completed\n");
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
log(1, "receive REGISTER\n");
# save location before initiating welcome
save("location");
log(1, "save ??\n");
record_route();
sl_send_reply("200", "ok");
# welcome message
log(1, "route3?? ?? \n");
route(3);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
log(1, "Usrloc DB?? ?? ? ?? \n");
break;
};
};
if(method=="BYE"){
log(1, "receive BYE\n");
record_route();
sl_send_reply("200", "ok");
break;
};
t_relay();
}
# welcome message -- if a REGISTER succeeded, look if it is the first-time;
# if so, initiate click-to-dial transaction to set up call to an
announcement
# server; edit the config values first to make it work
route[3] {
if (!exec_msg('
# config:
# --announcement server URI
ANS="sip:mijin@sam.cse"
# --SIP domain
# DOMAIN="192.168.178.193"
DOMAIN="192.168.178.194"
# ctd
CTD=${HOME}/sip_router/examples/ctd.sh
# ------------------------------------
# check if first time ...
SIP_UID=`echo $SIP_HF_TO | sed -e
"s/^.*sip:\([a-zA-Z0-9_\.]*\)@.*$/\1/g"`
QUERY="select flag from subscriber
where username=\"$SIP_UID\";
update subscriber set flag=\"x\"
where username=\"$SIP_UID\" ";
mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null
# ... if so, c-t-d to announcement server
if [ "$?" -ne 0 ] ; then
# flag was not set to x yet -- first-time registration;
# initiate a call from telephone of the user to an announcement server
$CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1
fi
')) {
log(1, "welcome exec failed\n");
}
log(1, "-------route3 ? \n");
}
_________________________________________________________________
???? ?? ??? ???? ???... MSN ??? ???
http://vod.msn.co.kr
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
I have this avpops test going...
avp_db_load("$ruri/username", "$urire/$urire_scheme");
# Load the urire column for this and check to see if
# it matches the $from address
if(avp_check("$urire", "eq/$from/I"))
{
do stuff
}
I'm calling from the number number(a)domain.com. If I use the above, the
avp_check test fails. If I put
if(avp_check("$urire", "eq/number(a)domain.com/I"))
instead, the check works exactly the way it's supposed to. It's almost as if
the $from address isn't right. Is there some way I can check what it's doing
at this point other than this painful trial and error?
N.
Hi!
I realy curious about the details of how midiaproxy system works.I mean you know there three parts proxy sever, proxy dispather and mediaproxy module as i know.I got some general ideas from the readme file.But I dont think thats enough.Please send me some detail files or maybe one example or something.Thanks very much!
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear....
I inserted command like this.
What I want is that
when I loaded test.cfg file, what I check SIP messages using sipp client
mode
I want 200 OK. But, as you see, I received "SIP/2.0 483 Too Many Hops"
error message.
So I was advanced about this.
- alias="sam.cse"//SIP_ DOMAIN
alias="server.sam.cse"//my_computer.SIP_DOMAIN
alias="192.168.178.194"//my_ip
- add my hostname in /etc/host
- change to : if (!mf_process_maxfwd_header("100")
but, I couldn't solve this problem.
I would really appreciate for giving some advise.
Thank you in advance.............................
mi-jin Song
=====================================================
[root@cse4 ~]# ser -f test.cfg
Listening on
udp: 127.0.0.1 [127.0.0.1]:5060
udp: 192.168.178.194 [192.168.178.194]:5060
tcp: 127.0.0.1 [127.0.0.1]:5060
tcp: 192.168.178.194 [192.168.178.194]:5060
Aliases:
tcp: server:5060
tcp: localhost:5060
tcp: localhost.localdomain:5060
tcp: cse4:5060
udp: server:5060
udp: localhost:5060
udp: localhost.localdomain:5060
udp: cse4:5060
*: 192.168.178.194:*
*: sam.cse:*
*: server.sam.cse:*
stateless - initializing
0(0) Maxfwd module- initializing
exec - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) ERROR: tcp_init: bind(6, 0x810dafc, 16) on 127.0.0.1: Address already
in use
=====================================================
[root@cse4 sipp]# ./sipp -sn uac 192.168.178.194
Resolving 192.168.178.194...
------------------------------ Scenario Screen -------- [1-4]: Change
Screen --
Call-rate(length) Port Total-time Total-calls Remote-host
10.0(0 ms)/1.000s 5061 2.54 s 25
192.168.178.194:5060(UDP)
5 new calls during 0.538 s period 2 ms scheduler resolution
0 concurrent calls (limit 30) Peak was 1 calls, after 0 s
0 out-of-call msg (discarded)
1 open sockets
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 25 0 0
100 <---------- 25 0 25
180 <---------- 0 0 0
200 <---------- E-RTD 0 0 0
ACK ----------> 0 0
[ 0 ms]
BYE ----------> 0 0 0
200 <---------- 0 0 0
------------------------------ Test Terminated
---------------------------------
----------------------------- Statistics Screen ------- [1-4]: Change
Screen --
Start Time | 2005-09-23 20:03:13
Last Reset Time | 2005-09-23 20:03:15
Current Time | 2005-09-23 20:03:16
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:536 | 00:00:02:543
Call Rate | 9.328 cps | 9.831 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 5 | 25
Total Call created | | 25
Current Call | 0 |
-------------------------+---------------------------+--------------------------
Successful call | 0 | 0
Failed call | 6 | 25
-------------------------+---------------------------+--------------------------
Response Time | 00:00:00:000 | 00:00:00:000
Call Length | 00:00:00:004 | 00:00:00:004
------------------------------ Test Terminated
--------------------------------
2005-09-23 20:03:16: Unexpected message for Call-ID
'25.4743.127.0.0.1(a)sipp.call.id': while expecting '100' response, received
'SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP 127.0.0.1:5061;received=192.168.178.194
From: sipp <sip:sipp@127.0.0.1:5061>;tag=25
To: sut
<sip:service@192.168.178.194:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6e68
Call-ID: 25.4743.127.0.0.1(a)sipp.call.id
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=4688
req_src_ip=192.168.178.194 req_src_port=5060
in_uri=sip:service@192.168.178.194:5060
out_uri=sip:service@192.168.178.194:5060 via_cnt==18"
' .
sipp: There were more errors, enable -trace_err to log them.
=======================================================
<< test.cfg >>
#
# $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $
#
# welcome message for new subscribers; based on exec
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
# must be yes since REGISTER processing causes an INVITE to be sent,
# which needs to be processed by another process
fork=yes
children=4
alias="server.sam.cse"
alias="sam.cse"
alias="192.168.178.194"
# debugging
log_stderror=yes # (cmd line: -E)
mhomed=yes
fifo="/tmp/ser_fifo"
# ------------------ 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/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/exec.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" )
modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser")
# ------------------------- request routing logic -------------------
# main routing logic
route{
log(1, "\n\n");
log(1, "START!!!\n");
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("15")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len>=204822) {
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=="INVITE") record_route(); # 1=loose routing
if(method=="INVITE"){
log(1, "receive INVITE\n");
record_route();
sl_send_reply("200", "ok");
};
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if(method=="ACK"){
# log(1, "receive ACK\n");
# };
log(1, "record-route processing completed\n");
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
log(1, "receive REGISTER\n");
# save location before initiating welcome
save("location");
log(1, "save 성공\n");
record_route();
sl_send_reply("200", "ok");
# welcome message
log(1, "route3으로 이동 \n");
route(3);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
log(1, "Usrloc DB에서 찾을 수 없음 \n");
break;
};
};
if(method=="BYE"){
log(1, "receive BYE\n");
record_route();
sl_send_reply("200", "ok");
break;
};
t_relay();
}
# welcome message -- if a REGISTER succeeded, look if it is the first-time;
# if so, initiate click-to-dial transaction to set up call to an
announcement
# server; edit the config values first to make it work
route[3] {
if (!exec_msg('
# config:
# --announcement server URI
ANS="sip:mijin@sam.cse"
# --SIP domain
# DOMAIN="192.168.178.193"
DOMAIN="192.168.178.194"
# ctd
CTD=${HOME}/sip_router/examples/ctd.sh
# ------------------------------------
# check if first time ...
SIP_UID=`echo $SIP_HF_TO | sed -e
"s/^.*sip:\([a-zA-Z0-9_\.]*\)@.*$/\1/g"`
QUERY="select flag from subscriber
where username=\"$SIP_UID\";
update subscriber set flag=\"x\"
where username=\"$SIP_UID\" ";
mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null
# ... if so, c-t-d to announcement server
if [ "$?" -ne 0 ] ; then
# flag was not set to x yet -- first-time registration;
# initiate a call from telephone of the user to an announcement server
$CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1
fi
')) {
log(1, "welcome exec failed\n");
}
log(1, "-------route3 끝 \n");
}
_________________________________________________________________
책상위에 다리 올리고 느긋하게 즐긴다... MSN 온라인 상영관
http://vod.msn.co.kr
Hi, I has send a MESSAGE to my SER server and it allways send me the
response "513 message to big".
I send it with sipsak -s sip:ninguna@10.22.1.249 -vvv -l 5060 -M, when
I send a REGISTER or INVITE it works whell.
I have alias for various domains, an it can´t be te error, have somo
one had this error ever? How can I get RIP of it? please help.
hi, All
Good morning
Can anybody tell me what is a exact difference between Class4 and Class5 softswitch ?
Because many people or company have there own defination
Regards
--Manoj
---------------------------------
Yahoo! for Good
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confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear. Alll
Hi, I have some questions.............
My study object is "communication among many computers(three
computers)based SIP".
So I have used SER with sipsak. (operating system : Linux Fedora 3)
but SER was not runned well.
In frankly, I don't know what to do.
Following is my try,
my SIP_DOMAIN is "sam.cse" and ip is 192.168.178.194.
[root@cse4 ~]# sipsak -s sip:mijin@sam.cse -M
warning: ignoring -i option when in usrloc mode
received:
SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP
127.0.0.1:32778;received=192.168.178.194;branch=z9hG4bK.01b7ca3a;rport=32778
From: sip:sipsak@127.0.0.1:32778;tag=141698a0
To: sip:mijin@sam.cse;tag=b27e1a1d33761e85846fc98f5f3a7e58.5de2
Call-ID: 337025184(a)127.0.0.1
CSeq: 1 MESSAGE
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=3626
req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:mijin@sam.cse
out_uri=sip:mijin@sam.cse via_cnt==18"
error: did not received the 'MESSAGE' that was sent (see above). aborting
===============================================
I want to communicate another computer has ip:192.168.178.193.
[root@cse4 ~]# ser -f /usr/local/etc/ser/ser.cfg -l 192.168.178.193
0(3965) WARNING: fix_socket_list: could not rev. resolve 192.168.178.193
0(3965) WARNING: fix_socket_list: could not rev. resolve 192.168.178.193
Listening on
udp: 192.168.178.193 [192.168.178.193]:5060
tcp: 192.168.178.193 [192.168.178.193]:5060
Aliases:
stateless - initializing
0(0) Maxfwd module- initializing
textops - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) ERROR: udp_init: bind(4, 0x810ab6c, 16) on 192.168.178.193: Cannot
assign requested address
===============================================
I setted my hostname and ip in /etc/host.
and inserted
alias="sam.cse"
alias="server.sam.cse"
alias="192.168.178.194"
but, as you see, I received "483 Too Many Hops" message;;;;;
I don't know what to do................
I appreciate if you give me some solutions, please ^^
Thank you in advance....
mi-jin Song
_________________________________________________________________
증권 정보 가장 빠르고 편하게 보실 수 있습니다. MSN 증권/투자
http://www.msn.co.kr/stock/
Hi,
I have a question on how to enter values specifically ip_address in the gw
table?
I would like to give a go this LCR module. I'm facing a problem where by: -
1. I created the tables (gw, lcr, gw_grp)
2. Inserting values into the tables ready for use.
The gw:ip_addr field is int data type, not able to store the decimal IP
format!!
How to manage that?
Thanks in advance.
