Hi all
Maybe a very simple question - but I cannot see the solution - I am more or less newbie ;)
I need to do some string operations in kamailio.cfg.
How can I get the cli from the following string:
sip:+41523940347@195.216.67.103;user=phone
I only would need +41523940347 in a variable
Something like
$avp(s:myCli) = ^sip:\+(\d{11})@.{1,40}$
Any idea?
Thanks in advance
Oli
Hi Daniel,
Issue got solved. I am just sharing it.
My requirement is I have to parse the SDP from cfg which is coming in 200
OKEY(Initiated by UAC once the called party answers the call).
On reply route we can capture the responses for those requests which we set
t_on_reply();(externally we have to call for each request).
But reply route we can capture all replies. So that we can parse our
required headers or msg body.
Regards
Surendra
******************************
Hi!
I have question concerning TLS module. I would like to enable TLS 1.2 protocol.
As I understand the TLS module is compiled with openssl, which as far as i know has TLS 1.2 support since openssl 1.0.1.
I am wondering if there is way to enable TLS 1.2 on the TLS module?
Many thanks
Marc
Hi,
I have kamailio server and a freeswitch server during a call the kamailio
passes the call to freeswitch to play a media and back to kamailio for
routing to the destination, this works fine however for samsung devices
(any ROM using linphone). On other device brands, it does not work it
always returns USER_BUSY or NUMBER_NOT_FOUND. After investigation, I found
out that if the transport is UDP then it works on all phones, the problem
happens if I use TCP as the transport. Anyone had experienced on this?
Thanks,
Jason
Please post your questions to the mailing list instead of sending
private e-mails.
The subscriber provisioning via xhttp is supported only for plaintext
passwords. Set
modparam("auth_db", "calculate_ha1", 1)
and it will work fine.
Regards,
Ovidiu Sas
On Sat, Nov 2, 2013 at 12:22 PM, Abdul Hakeem <alhakeem(a)gmail.com> wrote:
> Hello,
>
> I just want to enquire if the subscriber full provisioning is ready.
> Best regards,
> Abdul Hakeem
>
>
--
VoIP Embedded, Inc.
http://www.voipembedded.com
kamailio dispatcher module in my environment uses only UDP
I have not been able to find how I can configure kamailio dispatcher module to use TCP
Would appreciate any help
Bijan
Hi all,
I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is
used.
I have the system running, lines registered in Kamailio are able to call
to several destinations, but I have a problem with one provider that use
Asymmetric RTP (others providers use Symmetric RTP and I dont have any
issues).
A -----> Kamailio & RTPProxy ----------> Trunk -------> Provider
(Asymmetric RTP) -------> B
A is calling B.
The problem is that line in kamailio A can listen to B, but B can not
listen what A says.
I have captured the sip dialog and rtp traffic and I can see that:
Kamailio tells to provider that is using the port 52388 (media port in
SDP).
Providers tells Kamailio that is using the port 5394 (media port in
SDP).
RTP traffic from Providers to kamailio goes from port 5392 to 52388.
RTP traffic from Kamailio to Provider goes from port 52388 to 5392.
I have not found an RFC regarding Asymmetric RTP. The question are,
why RTP is sent from Kamailio (rtpproxy) to Provider to destination
port 5392?
It should use 5394 as SDP indicate?
or, port 5392 is learned when RTP traffic was received from 5392 to
52388?
# kamailio -V
version: kamailio 3.3.0 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 22:17:07 Jun 18 2012 with gcc 4.4.5
# rtpproxy -v
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup
command
Extension 20081224: Support for session timeout notifications
I hope to be clear with this explanation.
Regards,
Lucas Girard
Hi!
I have a simple msilo setup which is working fine:
-If the user is registered, the messages are delivered right away
-If the user is not registered, the messages are stored in msilo and got deliverd at the next register
Here is the relevant part of the cfg:
---------------
if (is_method("MESSAGE"))
{
log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n");
route(MSILO_MESSAGE);
}
.
.
.
route[MSILO_MESSAGE]{
if (lookup("location")){
t_relay();
}else{
m_store("$ru");
sl_send_reply("200", "OK");
}
exit;
}
.
.
.
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
# MSILO - dumping user's offline messages
m_dump();
exit;
}
}
-------------------
Thats working fine.
Now, we have a use case where a user is registered, but got terminated in between abd not reachable.
So the user is in the userloc table, but not reachable.
In this case ,since the user is still in userloc, the t_relay() will be executed, and it drops a 408 (timeout) code which is fine. However, the process end here. The message is then dropped.
What i would like to achieve is to make sure that t_relay() was executed with 200, otherwise store the message and deliver it at the next register.
I was trying this:
------------
##storing offline messages (new)
route[MSILO_MESSAGE]{
if (lookup("location")){
if (!t_relay()) {
m_store("$ru");
sl_send_reply("200", "OK");
}
exit;
}
else{
m_store("$ru");
sl_send_reply("200", "OK");
}
exit;
}
--------
but i have still the same behaviour.
Can you help me, what do I wrong?
Many thnaks
Marc. M
Hi,
what do you mean with "capture from cfg file" ? Do you want to log 200OK message in kamailio log file ?
Daniel
On Friday, November 1, 2013 08:36 CET, "Surendra" <surendra.pullaiah(a)plintron.com> wrote:
> Hi all,
>
>
>
> I have a requirement , i.e. I need to capture 200 ok(with
> SDP) from cfg file. This 200 okey will be getting once the called party
> answers the call.
>
> Can anyone help me to get out of this.
>
>
>
> Regards
>
> Surendra
>
>
>
>
>
Hi all,
I have a requirement , i.e. I need to capture 200 ok(with
SDP) from cfg file. This 200 okey will be getting once the called party
answers the call.
Can anyone help me to get out of this.
Regards
Surendra
