Hi
I have just install 1.5 and I am trying to setup TLS
I ran
kamctl tls rootCA
and it created ok
However the docs at http://www.kamailio.org/docs/tls.htm next say to run
kamailioctl tls userCERT
However I don't have a kamailioctl on my system.
Any ideas anyone?
Regards
Jon
--
Jon Farmer
Tel: 07795 118140
Email: viperdudeuk(a)gmail.com
Twitter: @viperdudeuk
Hi,
Is it possible to add an header reason in a cancel?
We receive a CANCEL without header reason and we need to forward to
another SIP proxy with the reason header?
We use this code in my conf:
if(is_method("CANCEL"))
{
if(!is_present_hf("Reason:"))
{
xlog("L_ERR", "reason missing");
append_hf("reason: Q.850; cause=31\r\n");
xlog("L_ERR", "add raison");
}
}
Thanks,
regards,
--
Alexandre Rendour
Acropolis Telecom <http://www.acropolistelecom.net>
Direct: +33 (0) 181813201
Support: +33 (0) 811 851 851
rendour(a)acropolistelecom.net
<mailto:rendour@acropolistelecom.net>
Adresse : 161-163 avenue Gallieni
Paris - Porte de Bagnolet
93170 Bagnolet
I have the bellow setup for openser, yet this wont work in Kamilio.
Wondering if anyone could help to make this routing code work?
What Im looking to do is receive Invites in UDP and relaying it via TCP....
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
};
route(1);
}
route[1]
{
if (src_ip==10.98.6.5) {
if (dst_port==5060) {
t_relay("tcp:10.98.118.20:5060");
}
if (dst_port==5065) {
t_relay("tcp:10.98.118.20:5065");
}
else if (dst_port==5066) {
t_relay("tcp:10.98.118.20:5066");
}
else if (dst_port==5067) {
t_relay("tcp:10.98.118.20:5067");
}
# else {
# t_relay("tcp:10.98.118.20:5060");
# }
}
else {
t_relay("udp:10.98.6.5:5060");
};
}
Nelson Pereira
Senior Network Specialist
Protus<http://www.protus.com/>
npereira(a)protus.com<mailto:name@protus.com>
phone: 613.733.0000 ext.528
MyFax: 613.822.5083
MyFax.com<http://www.myfax.com/> | my1voice.com<http://www.my1voice.com/> | Campaigner.com<http://www.campaigner.com/>
Good afternoon,
I'm trying to configure permissions module to allow calls from an IP
address. I've the module loaded, the entry on the data base table
trusted. But according to the examples I've seen on the list and other
sources, the call to allow_trusted() has to be placed just before the
INVITE authorization section like:
if (is_from_local()){
if(!allow_trusted()){
...authorization...
In my scenario allow_tusted() is never reached and thus Kamailo allows
the call regardless the table trusted is populated with the the allowed
proxy IP or not.
The INVITEs that I want to allow come from another proxy and then I
understand is_from_local() is always evaluated false for those messages.
And allow_trusted isn't reached.
Where on the script should the condition to allow only the calls from
the external proxy IP be placed?
Thanks in advance.
Vicente.
Hi,
First I'm a complete SIP newbie.
We've purchased a new pabx which is a complete piece of crap (Alcatel
OmniPCX OXO) : no LDAP support, no registrar functionnality, and so
on... and it seems there's no way to upgrade.
This pabx is on a subnet different from the one used by desktop
computers.
What I'd like to achieve is to be able to install SIP soft phones onto
desktop computers (like ekiga for example), allow them to call all
non-IP phones already configured in the pabx, and be called by any
non-IP phone through the pabx.
Since the pabx doesn't have a registrar mode, it can't associate IP
addresses with phone numbers. So in ekiga, I can configure the pabx as a
SIP proxy, and my ekiga can call any normal (non-IP) phone connected to
the pabx. This works fine, but is only one half of what I want, since I
want these SIP phones to be callable from non-IP phones.
Enters openser, which I plan to use as a registrar and router.
So I've installed openser 1.3.2 onto a Debian Squeeze box. I can
successfully register SIP soft phones to it, and they can call each
other just fine calling (number(a)sip.example.com). This openser box is on
the same subnet as the Alcatel pabx (so different from the one used by
SIP soft phones).
All our "normal" phone numbers are 6 digits long and begin with 290,
so I want to route all call numbers beginning with "290" to the Alcatel
pabx.
So I've modified the default openser.cfg to contain :
--- CUT ---
port=5060
listen=udp:10.125.0.1:5060
...
route {
...
if (is_method("INVITE")) {
setflag(1);
if (uri=~"sip:290[0-9]{3}@sip.example.com") {
xlog("#### CALL TO NON-IP PHONE ####\n");
route(2);
}
}
...
}
...
# route[1] was in original openser.cfg
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
#
# I've added the following
route[2] {
rewritehostport("pabx-alcatel.example.com:5060");
xlog("#### Server and port rewritten ####\n");
route(1);
}...
--- CUT ---
Now in ekiga if I dial sip:number@pabx-alcatel.example.com:5060 it works
fine as it always did, because this call is direct.
However if I dial sip:number@sip.example.com:5060 the host and port in
URL are correctly rewritten in sip:number@pabx-alcatel.example.com:5060
but nothing happens : the call doesn't ring and fails immediately.
In the logs when dialing sip:290081@sip.example.com:5060 I've got this,
so I suppose it *SHOULD* work :
--- CUT ---
/usr/sbin/openser[759]: #### CALL TO NON-IP PHONE ####
/usr/sbin/openser[759]: #### Server and port rewritten ####
/usr/sbin/openser[759]: new branch at sip:290081@pabx-alcatel.example.com:5060
/usr/sbin/openser[760]: incoming reply
/usr/sbin/openser[758]: incoming reply
--- CUT ---
but it doesn't work as expected.
any idea why ?
NB : I understand this would only solve half of my problem, the other
part will be solved (I hope) with the help of the guy who installed the
pabx.
Thanks in advance
Jerome Alet
Hi,
I'm having some challenges with Siremis, and it says on the project
website that this list is the place to ask - so here are the two
issues:
1. Repeats:
I had problems with repeats in the cdrs table. This was possible,
because I had forgotten to set the correct unique index (call_id +
from_tag + to_tag). Once this was set, it was obviously not possible
to make the repeats anymore - so problem solved. But I still wonder:
why was it trying to calculate calls that had already been calculated?
Shouldn't it ignore these and only treat new ones?
2. Missing calcs
After an unsuccessful attempt to upgrade MySql from 5.0 to 5.1, it
seems that not all calls are being calculated and put in the cdrs
table. The CRON job was stopped due to the failed upgrade of MySql,
but I started it again later - MySql was not down. However, I thought
it would 'catch up' with the transaction that had not been calculated
while the CRON job for the stored procedure had been off, but it did
not. In fact, it seems that not all calls are being calculated and put
inthe cdrs table - I'm missing almost half the calls. Any idea what
this can be?
Thanks!!
//Anders
Thanks for the response, although I'm not seing anything coming out when tail'ing the syslog file.
I only see when kamailio starts:
May 19 10:22:14 qa-qsuitePBX-01 kamailio: INFO: <core> [tcp_main.c:4150]: init_tcp: using epoll_lt as the io watch method (auto detected)
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: usrloc [hslot.c:53]: locks array size 512
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142
May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1804]: INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config)
Do I need anything else to enable?
Nelson Pereira
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: Wednesday, May 19, 2010 10:08 AM
To: Nelson Pereira
Subject: Re: [SR-Users] Kamailio 3.0.1 output debug to terminal
Kamailio default uses syslog logging.
I suggest to start a second console and use:
tail -f /var/log/syslog
you can redirect syslog to a dedicated file using, see:
http://www.kamailio.org/dokuwiki/doku.php/tutorials:debug-syslog-messages#c…
regards
klaus
Am 19.05.2010 15:50, schrieb Nelson Pereira:
> How do you output debug info to the terminal?
>
> I don't understand this :
>
> # Several features can be enabled using '#!define WITH_FEATURE' directives:
>
> #
>
> # *** To run in debug mode:
>
> # - define WITH_DEBUG
>
> How do I get debug and output to screen?
>
> *Nelson Pereira*
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
How do you output debug info to the terminal?
I don't understand this :
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
How do I get debug and output to screen?
Nelson Pereira
