Hi,
*
*
*
*
We are facing a problem in our opener proxy receiving: *"Message too big"
error.*
Looking at the INVITE message sent by opener I notice the following in the
record-route header:
Record-Route:
<sip:10.10.10.10;lr=on;ftag=5114A38-F80;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA->
Is this a valid value for vsd? Can we correct the value?
Thanks,
R
Hi, I'm using kamailio 3.2.3 and I use permissions.so
But when I reload Kamailio , error:
failed to initialize the allow_address function
0(3815) ERROR: <core> [sr_module.c:932]: init_mod(): Error while
initializing module permissions (/usr/lib/kamailio/modules_k/permissions.so)
ERROR: error while initializing modules
I don't use this function in kamailio.cfg.. just allow_trusted !!!
Any idea why this error still appears?
Best regards
--
*Grégoire Vandendeurpel, *
*
*
*IT Sector*
Hi all,
When record routing is enabled, that means, the proxy will act statefull mode,
There has to be a mechanism for storing dialogue states.
Like the location of the peers (Contact header filed of initial INVITE
and 200 OK).
And i suspect these states are stored in variables which has a lifespan same as
the dialogue itself. I guess important variables (Contact header,
local and remote tag etc)
are exported to configuration file also.
I'm aware of the dialogue module but i'm kind of convinced there has
to be a simpler way
for achieving my goal.
Our Sip gateway provider sending a wrong BYE when PSTN side phone hangs up.
Instead of using Contact Header of the initial INVITE, its sending
BYEs with its IP as ruri.
So the BYE message is looping between proxy and gateway instead of
relayed to UAC.
What i'm trying to achieve is following :
route[RELAY]
{
if(is_method('BYE'))
{
<rewrite ruri with contact field of the initial invite>
t_relay()
}
}
--
-aft
Hi,
I'm installing a brand-new Kamailio / Siremis system.
I have an issue with Siremis failing to load PDO extensions. I have
other systems working O.K. on Centos 5.x
This is my configuration:
Centos 6.2
php-5.3.3-3.el6_2.8.i686.rpm
php-cli-5.3.3-3.el6_2.8.i686.rpm
php-common-5.3.3-3.el6_2.8.i686.rpm
php-mysql-5.3.3-3.el6_2.8.i686.rpm
php-pdo-5.3.3-3.el6_2.8.i686.rpm
Siremis 3.2.1
php -m
[PHP Modules]
bz2
calendar
Core
ctype
curl
date
ereg
exif
fileinfo
filter
ftp
gettext
gmp
hash
iconv
json
libxml
mysql
mysqli
openssl
pcntl
pcre
PDO
pdo_mysql
pdo_sqlite
Phar
readline
Reflection
session
shmop
SimpleXML
sockets
SPL
sqlite3
standard
tokenizer
xml
zip
zlib
[Zend Modules]
The area where it screws up is in util.php while I am setting up Siremis
via web-page
$status[5]['item'] = 'PDO extensions';
$pdos = array();
if (extension_loaded('pdo')) $pdos[] = "pdo";
if (extension_loaded('pdo_mysql')) $pdos[] = "pdo_mysql";
if (extension_loaded('pdo_mssql')) $pdos[] = "pdo_mssql";
if (extension_loaded('pdo_oci')) $pdos[] = "pdo_oci";
if (extension_loaded('pdo_pgsql')) $pdos[] = "pdo_pgsql";
$status[5]['value'] = implode(", ", $pdos);
$status[5]['status'] = $pdos[0]=='pdo' ? 'OK' : 'FAIL - PDO
extensions are required.';
I have commented out the test but it fails later saying the PDO
extensions are not loaded.
I note that the php modules report gives upper case but the test is
lower case.
(I understand the extension_loaded function is case insensitive)
Any assistance appreciated.
Hello,
I'm working on a residential type application where we are using Kamailio for NAT traversal and Freeswitch as a voicemail and media server. When a UA that is behind NAT sends an INVITE to check voicemail everything works correctly until the user listens to the message. The sdp in the initial INVITE is rewritten and rtp proxy is working but Freeswitch (on a public IP) then sends an UPDATE to display the caller name of the person who left the message. The problem is that the UAC (in this case a Polycom phone) then responds with its private IP in the SDP. Is there a was to handle these UPDATEs? I'm using Kamailio 3.2.3 with a fairly stock config. This is an excerpt of the config file with the NAT handling route:
# RTPProxy control
route[NATMANAGE] {
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
return;
}
Thanks,
Spencer
Hi:
I'm trying to configure a Kamailio 3.2 server as a Redirect Server.
A softswitch we own will send a SIP INVITE for some calls like this "INVITE
sip:102@sipwise.telecentro.net.ar SIP/2.0".
Once detected the B number "102" (in this example), the kamailio server
will search in a sqlite3 database the transltated number the call should be
directed.
And then the server will send a 300 Redirect with the new request uri. Like
this "SIP/2.0 300 Redirect /.../ Contact:
sip:63793266@voip.telecentro.net.ar".
And the softswitch will let the call go to the right number.
The translation desition is made based on the geographic area where the A
party is located.
If the translated number is like 63793266 or 1557311721 the system is
working ok. So the .cfg file seems to be OK.
But if the translated number is like 8003330303 the variable holding the
SQL return is set to "-486604289".
I think this is related to data types.
In the logs I can see that kamailio is defaulting datatypes to INT (I've
tested varchar, int, and bigint in sqlite3), but I don't know how large
could an integer be in Kamailio. According to sqlite3 documentation the
number 8003330303 is not long enough and in fact the number is in the
database.
Has anybody seen something like this?
Can anyone help?
I can provide the hole script and database if this helps.
Regards,
Sebastian Ferguson
Hi,
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So long,
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P.S.: I'll give a presentation about Kamailio, IMS and Rich
Communications on Wednesday, May 23rd, 12:45 - 13:30 (Room Berlin I).
--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
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Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
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--
Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the
date!
Hi,
I am using kamailio 3.2.2 and the uac_redirect module to evaluate the
response from a SIP redirect server.
My problem is, that kamailio does not pick the target with the highest
priority, but the one with the lowest.
In the 302 the kamailio is receiving, there is a contact header like
this:
Contact:<sip:+49xxx@a.a.a.a:5060;user=phone>;q=0.5,<sip:
+49xxx@b.b.b.b:5060;user=phone>;q=0.25.
In my failure_route block I basically use the following commands
get_redirects("1");
t_relay();
The call then gets redirected to b.b.b.b whereas I would expect it to go
to a.a.a.a
When I change my script to
get_redirects("*");
the call is forked to both IPs in parallel (starting with a.a.a.a)
Is this a bug in uac_redirect or am I using it in a wrong way?
Regards,
Stefan