Hello All!
Can somebody share a working/example config with biderectional
nathelper, tcp transport and push notifications (tsilo) for Kamailio
>=4.3?
Thanks in advance!
--
lifayk
e: lifayk(a)gmail.com
pgp: 0x84CAC40B
Hi,
The arch looks like this
SBC1 <-------- SIP-Router (Kamailio) <------------- Asterisk
Asterisk sends an invite to kamailio using "sip:" as request uri scheme,
but SBC1 expects that invite to arrive using tel:, is there a way to change
this in kamailio and forward that invite? Something like:
$ru = "tel: +" + $rU;
Thanks!!!
Hi,
I would like to enable and use sctp in kamailio but I do not know
exactly how to do it. It seems that documentation is a little bit
inconsistent.
1. I have Centos 7 and kamailio installed from repositories.
2. I set loadmodule "sctp" in kamailio.cfg
3. I got this error:
Jul 9 08:08:54 kamailio: : sctp [sctp_mod.c:99]: mod_register(): sctp
core support not enabled
Jul 9 08:08:54 kamailio: ERROR: <core> [sr_module.c:600]:
load_module(): /usr/lib64/kamailio/modules/sctp.so: mod_register
failed
Jul 9 08:08:54 cw-aio kamailio: : <core> [cfg.y:3432]: yyerror_at():
parse error in config file /etc/kamailio_pcscf/kamailio.cfg, line 230,
column 12-17: failed to load module
4. I checked mod_register() function and found out that USE_SCTP should be set.
5. kamailio -V did not show USE_SCTP
6. Next I tried to compile kamailio sources from 4.3 branch
make include_modules="db_mysql tls sctp" SCTP=1 cfg
make all
7. Still the same problem "mod_register(): sctp core support not enabled"
8. Then I added USE_SCTP to C_DEFS and compile again. This attempt
was successfull. I was able to see USE_SCTP in kamailio -V output.
Questions:
1. What is the right procedure to enable sctp in kamailio?
2. Why the -DUSE_SCTP was not used when SCTP=1 was used. According to
documentation:
"The core Makefile variable SCTP must be set to 1 (which is by default
set to 1 in Makefile.defs) and sources compiled with -DUSE_SCTP
(automatically set when SCTP=1). In other words, if core Makefiles are
not changed and SCTP variable is not overwritten from command line,
then the SCTP support in core is enabled."
Regards,
Peter
Hello,
I'm testing some basic flags with nat_uac_test. I have the following
(rudimentary) script:
route[NATDETECT] {
#!ifdef WITH_NAT
if (has_body("application/sdp")) {
if (nat_uac_test("18")) {
xlog("SCRIPT: SDP NAT Detected\n");
setflag(6);
}
} else {
if (nat_uac_test("2")) {
xlog("SCRIPT: Other NAT Detected\n");
setflag(6);
}
}
if (isflagset(6)) {
xlog("SCRIPT: NAT detected\n");
force_rport();
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop())
set_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}
When presented with this REGISTER message (ngrep output, so first
line is IP source address and port):
U 192.168.113.17:6060 -> 192.168.113.11:5060
REGISTER sip:192.168.113.11;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport.
From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40.
To: <sip:user_1@192.168.113.11>.
CSeq: 10 REGISTER.
Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40.
Max-Forwards: 70.
User-Agent: FreeSWITCH.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600.
Content-Length: 0.
.
U 192.168.113.11:5060 -> 192.168.113.17:6060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport=6060;received=192.168.113.17.
From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40.
To: <sip:user_1@192.168.113.11>;tag=b27e1a1d33761e85846fc98f5f3a7e58.1045.
CSeq: 10 REGISTER.
Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40.
Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600;received="sip:192.168.113.17:6060".
Server: Kamailio.
Content-Length: 0.
.
Log output:
Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>:
SCRIPT: Other NAT Detected
Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>:
SCRIPT: NAT detected
Log output, the presence of the added "received" parameter in the
returned Contact, etc show that NAT was detected. However, the IP
source and port match the one presented in the Via header.
What am I missing here?
Thanks!
--
Kristian Kielhofner
Hi All,
I am having difficulty getting pua_publish to issue a NOTIFY when I send
an XMLRPC command to generate a PUBLISH into the presence server.
I have been reading the pua_mi module docs, but I cannot see where I am
going wrong with the following xmlrpc command, please would someone
point me in the correct direction.
I have an XML file which is the basis of the XMLRPC request via curl, I
have attached it to this email for you to view, please would you take a
look and let me know where I am going wrong, I suspect it is my message
body as the pua module complains about the message body being NULL:
Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua
[add_events.c:125]: pres_process_body(): while parsing xml memory
Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua
[send_publish.c:650]: send_publish(): while processing body
Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua
[send_publish.c:652]: send_publish(): NULL body
Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua_mi
[mi_func.c:290]: mi_pua_publish(): sending publish failed
Kamailio verion is "version: kamailio 4.3.0 (x86_64/linux) c6aa95"
I can issue other XMLRPC commands, urloc lookups method lists and those
work, I just cant see where I am going wrong with this one.
Any pointers would be greatly appreciated.
Thanks
Would be nice if it would be possible to
- make timer of "default_timeout" reset to 0 when keep-alive reply is
received from remote part of call leg, so that the timeout expires
(triggering call termination) only if some call party gets offline (we
don't want just to limit the duration of normal call, even if call
parties don't send any new SIP requests);
OR
- make unresponsiveness of call parties to keep-alive messages
(controlled with ka_timer, ka_interval parameters) lead to some
action, like call termination.
Please advise if I miss something, and how to implement this better.
Thanks in advance.
--
Andrey Utkin
On Thursday 09 July 2015 17:46:22 Srujan Kumar K wrote:
> I am connecting my kamailio server to MSC for my solution which has both
> inbound and outbound.
...
> Kamailio server IP : 172.22.14.12
> MSC IP : 172.28.0.68
Why does you kamailio config bind to the msc ipadress?
Hello all,
Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio.
Especially creating partitions, DRBD devices and mount points.
Thanks in advance,
-Sid
"May the light be with you." ______________________________________________
Siddhardha Garige
www.luminepixels.com
Hai,
I am connecting my kamailio server to MSC for my solution which has both
inbound and outbound. The SIP trunk is not UP and we don't see any packets
transmitted between the 2 nodes.
We are also trying to load SIPI/SIPT module
Kamailio server IP : 172.22.14.12
MSC IP : 172.28.0.68
I have also attached configuration file with this mail. Requesting your help
on this.
Regards
~Srujan
