Hi,
I'm looking for a H323 - SIP signalling gateway which I can use to bridge
video calls between my H323 Gatekeeper and Kamailio system.
Has anyone done anything like this before? Are you able to suggest any
suitable h323 - SIP video gateway solutions I could use?
Thanks
--
View this message in context: http://www.nabble.com/H323---SIP-Signalling-gateway-for-Kamailio-tp25530990…
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
Hello everyone, i have some questions about kamailio.
After i install rtpproxy 0.3version, the kamailio server can't work any more. The following is the error message. Dose any one know the reason? dose the rtpproxy 0.3 change something that kamailio need?
Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module registrar
Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module textops
Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module uri_db
Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module siputils
Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module xlog
Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module acc
Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb
Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb
Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb
Sep 23 11:07:57 [4925] ERROR:core:build_req_buf_from_sip_req: out of pkg memory
Sep 23 11:07:57 [4925] ERROR:tm:print_uac_request: no more shm_mem
Sep 23 11:07:57 [4925] ERROR:tm:t_forward_nonack: failure to add branches
Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer
Sep 23 11:07:57 [4925] ERROR:tm:relay_reply: no more share memory
Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer
Is there a known bug in Kamailio 1.4.1 that causes comparisons like
this to be inaccurate?
get_profile_size("people", "$avp(s:person_id)", "$var(u_call_count)");
...
if($var(u_call_count) >= $avp(s:call_limit))
In this case, $var(u_call_count) was 3 and $avp(s:call_limit) was 26 -
I printed both values beforehand. Yet, the comparison was evaluated
as true.
I worked around the problem by storing the profile size in an AVP as
well and comparing them instead, and it seems to have solved the problem:
if($avp(s:u_call_count) >= $avp(s:call_limit))
However, I was wondering if this is a known problem in 1.4.1 and
possibly 1.4.x.
Thanks!
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Hi, I've experimenting an absurd issue when a softswitch calls to
Kamailio and receives two 180 (with different To tags) due to parallel
forking made by Kamailio.
When the softswitch receives the second 180 (with the second To tag)
it sends a ridiculous CANCEL with To tag matching the second To tag
received (yes, a CANCEL with To tag !!!).
Of course Kamailio replies "403 Why do you send a CANCEL with To tag?
please read the RFC 3261". Unfortunatelly the softswitch didn't go to
the school and ignores the 403 by repeating the CANCEL. And the worst,
when *any* of the ringing phones answerw, the softswitch ignores the
200 so the call is never established.
IMHO there is nothing I can do to "fix" this stupid issue. I've tried
to reply 200 for the CANCEL but the softswitch is not happy (it seems
that it expects the 487 from the canceled branch). And of course, if I
do "t_relay()" for the CANCEL then Kamailio cancels both branches (the
correct behavior).
The only hyper exotic "solution" I can imagine is the following:
- Kamailio receives the wrong CANCEL with To tag "2".
- Reply 200 for the CANCEL to make the softswitch happy.
- Reply 487 with To tag "2" to make the softswitch happy (IMHO this is
not possible).
- Generate a spoofed BYE for the second branch (that with To tag "2").
Of course, this is not possible with dialog module as it doesn't
handle parallel forking.
Any suggestion? (yes, I would like to kill the softswitch).
Thanks.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
Hello,
I just committed a new function in modules_k/textops -
msg_apply_changes(). Hopefully the name is suggestive enough - the
function applies the changes done to the content of sip message - e.g.,
if you added a new header, once you use the function, that header will
become visible in config file.
Therefore the function tries to overcome the overhead of looping various
messages back to sip server just because modifications were not visible
once done. Another benefit is ability to access latest value for a field
(being header or body content) in case of need to hashes or signatures
for security purposes.
The function can be used only in request route for now. Be carefully
when using it, since it changes the expectation you had so far of always
working on initially received sip message. Once the function is sued,
initial content is lost forever. Since it builds a new buffer and
re-parses, it is not very recommended to use it extensively.
Example of usage:
append_hf("My-Header: yes\r\n");
if(msg_apply_changes())
{
# msg buffer has a new content
if(is_present_hf("My-Header"))
{
# will get always here
}
}
Testing and feedback is welcome!
Cheers,
Daniel
Hello everybody:
I've have some questions about tls connection.
How to build the tls connection between two proxy? there's just one function is_peer_verified() in tlsops module. Is it that we set the paramer, the proxy will connect by itself or we should build the connection in the config file explicitly? could someone give me a example?
Proxy1<----TLS--->Proxy2
Dear Experts,
I have been trying to configure IMC for quite
some time now..when ti try to create the chatroom (using my xlite
phone), using the command #create q, i get the following error.
*ERROR:imc:imc_handle_message: room [q] does not exist!
Sep 17 10:25:36 [2562] ERROR:imc:imc_manager: failed to handle 'message'
*
Iam posting my cfg below.. Your sujjestion would be of immense help to me..
#
# $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $
#
# Kamailio (OpenSER) SIP Server - basic configuration script
# - web: http://www.kamailio.org
# - svn: http://openser.svn.sourceforge.net/viewvc/openser/
#
# Direct your questions about this file to: <users(a)lists.kamailio.org>
debug=3
log_stderror=yes
log_facility=LOG_LOCAL0
fork=yes
children=4
port=5060
/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all
available) */
#listen=udp:192.168.1.2:5060
####### Modules Section ########
#set module path
mpath="/usr/local/lib/kamailio/modules/"
/* uncomment next line for MySQL DB support */
loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "uri_db.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "acc.so"
loadmodule "imc.so"
/* uncomment next lines for MySQL based authentication support
NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
NOTE: a DB (like db_mysql) module must be also loaded
NOTE: be sure and enable multi-domain support in all used modules
(see "multi-module params" section ) */
#loadmodule "domain.so"
/* uncomment the next two lines for presence server support
NOTE: a DB (like db_mysql) module must be also loaded */
#p#loadmodule "presence.so"
#p#loadmodule "presence_xml.so"
#n#loadmodule "nathelper.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)
# ----- rr params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# ----- uri_db params -----
/* by default we disable the DB support in the module as we do not need it
in this configuration */
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")
# ----- acc params -----
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
# ----- auth_db params -----
/* uncomment the following lines if you want to enable the DB based
authentication */
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("auth_db", "load_credentials", "")
#-------tm params-----
modparam("tm", "fr_timer", 10 )
modparam("tm", "fr_inv_timer", 15 )
modparam("tm", "wt_timer", 10 )
#-------imc params-----
modparam("imc", "db_url", "mysql://openser:openserrw@localhost/openser")
modparam("imc", "hash_size", 8)
modparam("imc", "imc_cmd_start_char", "#")
#n#modparam("usrloc", "nat_bflag", 6)
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
# NAT detection
route(4);
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(1);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(2);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK
after a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore and
discard.\n");
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(3);
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself)
/* replace with following line if multi-domain support is used */
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
##if($rd=="tls_domain1.net") {
## t_relay("tls:domain1.net");
## exit;
##} else if($rd=="tls_domain2.net") {
## t_relay("tls:domain2.net");
## exit;
##}
route(1);
}
# requests for my domain
if( is_method("PUBLISH|SUBSCRIBE"))
route(2);
if (is_method("REGISTER"))
{
if (!save("location"))
sl_reply_error();
exit;
}
#if(uri=~"sip:q.*@")
if(uri=~"sip:q@192.168.64.75")
{
# IMC - message
xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n");
if(is_method("MESSAGE"))
{
log("MESSAGE received -> processing with imc\n");
sl_send_reply("200", "ok");
imc_manager();
exit;
} else {
sl_send_reply("404", "not found");
exit;
};
} else {
xdbg("script: message not for room, from [$fu] r-uri [$ru]
msg [$rb]\n");
if(!lookup("location"))
{
sl_send_reply("404", "not found");
exit;
}
t_relay();
#sl_send_reply("200", "ok");
exit();
};
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
/* example how to enable some additional event routes */
if (is_method("INVITE")) {
#t_on_branch("1");
t_on_reply("1");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence server */
route[2]
{
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
/* uncomment the whole following route for enabling authentication */
route[3] {
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if ($au!=$tU)
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
} else {
# authenticate if from local subscriber (uncomment to enable auth)
if (from_uri==myself)
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (is_method("PUBLISH"))
{
if ($au!=$tU) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
} else {
if ($au!=$fU) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
}
consume_credentials();
# caller authenticated
}
}
return;
}
# Caller NAT detection route
/* uncomment the whole following route for enabling Caller NAT Detection */
route[4]{
return;
}
# RTPProxy control
/* uncomment the whole following route for enabling RTPProxy Control */
route[5] {
return;
}
branch_route[1] {
xdbg("new branch at $ru\n");
}
onreply_route[1] {
xdbg("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
-
Regards,
Rajesh
Hi to all,
i'm using openser-1.2.2 and i'm getting the error:
"Error while converting IP address string to in_addr"
while using the allow_address function.
I tried both with:
allow_address("$(ru{uri.host})", "$rp")
allow_address("$rd", "$rp")
I made debugging printng content of both $rd and $(ru{uri.host}) and
both return the correct value.
Don't know anymore where to search about it.
Any help?
Thank you,
regards,
Marcello
when i create account in x-lite,then i found error code like this:
Registration error:500 - Server Internal Error
i'm using SIP server with SER 0.9.6.
what's the problem with x-lite?
thanks,,