Hi,
Is it possible to save user location data in a MongoDB database?
If not, are you guys using p_usrloc in your projects? Can you please give me
some feedback?
Thanks,
Mickael
Hello all,
we are are a small network-provider and selling internet via FTTC and
FTTH. To complete the portfolio we also sell telephony via a rented
ClassV Service with provisioning API, but its not our system and not in
our network.
To get more informations about Registration-Status and calls I have now
installed a kamailio as a statefull proxy between the subscriber and the
softwitch:
|----------|
|subscriber|\ ---------
|----------| \ /| Mobile|
\ -------------- / ---------
|----------| \ ------------ | Class V | / --------
|subscriber|---------| Kamailio |--------| Softswitch |-------| PSTN |
|----------| / ------------ | | \ --------
/ -------------- \ ---------
|----------| / \| World |
|subscriber|/ ---------
|----------|
I am really new in kamailio, so I just use the configuration named "Edge
Proxy Configuration" shown in the documentation to the outbound module,
see http://kamailio.org/docs/modules/4.2.x/modules/outbound.html
example 1.1. We dont need NAT and STUN because all our subscribers get
official IPv4 addresses.
Outgoing calls (subscriber -> PSTN) are working fine, all SIP messages
pass the Kamailio Proxy. But incoming Calls (World -> subscriber) miss
the proxy and will be routed directly from the Softwitch to the subscriber.
I think because of the Contact-Header in the subscriber REGISTER:
Contact: <sip:49xxxxxxyyyyyy@134.xx.yy.zz;uniq=E41120B46C92D3818E72B>
This is the IP of the subscriber FritzBox.
So I just added some lines to replace the Contact-Header in the REGISTER
to get all incoming Calls from the Softswitch to the Kamailio:
if (is_method("REGISTER")) {
if (src_ip==CUSTOMER_NET) {
remove_hf("Route");
add_path();
$du = "sip:REGISTRAR_IP:REGISTRAR_PORT";
if ($ct != $null) {
subst('/^Contact:(.*)@[a-zA-Z0-9.]+(.*)$/Contact:\1@KAMAILIO_IP\2/ig');
}
}
}
and now the Contact-Header looks better:
Contact: <sip:49xxxxxxyyyyyy@KAMAILIO_IP;uniq=E41120B46C92D3818E72B>
Now the subscriber is registered on the Softswitch with the IP from the
Kamailio. And now the Kamailio get the incoming Call, but directly
route it back to the Softwitch and the call failed.
Now my questions:
- Is the "Edge Proxy Configuration" right for this setup?
- How the Kamailio keep all registered IPs and Lines to route an
incoming Call to the right subscriber?
- Why the Kamailio route an incoming call directly back to the
Softswitch?
Thanks a lot.
Regards
Thomas
Hi All
I am in need of being able to see what the duration of the call was at the
time of hang out.
I tried turning on the dialog module, but the result is that the values I
need are either null or always show as zero.
I tried to follow the suggestions in the thread about this here:
http://lists.sip-router.org/pipermail/sr-users/2010-October/065889.html
In the end my config looks like this:
loadmodule "dialog.so"
...
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "dlg_match_mode", 1)
...
request_route {
if (is_method("INVITE") && (! has_totag() ) ) {
dlg_manage();
}
if (is_method("BYE")) {
$var(elapsed) = ( $TV(s) - $dlg(start_ts) );
xlog ("L_INFO", "request_route DIALOG TEST: Completed $dlg(from_uri) to
$dlg(to_uri), elapsed: $var(elapsed), now seconds: $TV(s), dlg start time:
$dlg(start_ts), DLG_lifetime: $DLG_lifetime");
}
....
Note that I put at the top of the request route just for testing purposes
The result I get in the log is this:
INFO: <script>: request_route DIALOG TEST: Completed
sip:willf1976Test@develop-sbc.switchsoft.com to
sip:+18054515526@develop-sbc.switchsoft.com, elapsed: 1421305210, now
seconds: 1421305210, dlg start time: 0, DLG_lifetime: <null>
I was wondering if any one could point out to me what I have set up wrong.
Thanks in advance
All the best.
Will Ferrer
Switchsoft
Hi All,
i'm trying to get GEO IP Location for an IPv6 Setup.
We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in
modparams of geoip.
If i now place a call from an V6 User and do something like:
if(geoip_match("$si", "src")) {
xlog("L_NOTICE", "Call comes from IP '$si' ($gip(src=>cc))
CID=$ci F=$fU URI=$ru\n");
append_hf("X-GeoIP: $gip(src=>cc)\r\n");
}
it will not match for any reason
if i xlog the src IP it look like: "Src IP = 2A01:abc:321:123:0:0:0:2" so
it should match to the GEO IP Database.
In CSV version of geo IP database there is an entry for my IP Address.
Since the Module is not that verbose, i cannot tell whether the entry
exists or not. Just the result in my "if" is False.
Has anybody experience with GEOIP and IPv6?
thanks
Best regards,
Jöran
Hi,
I need to request to kamailio the list of all the active dialog (with some
dialog variable).
Now , i'm doing it with a program that call the proxy with the xmlrpc
module. The problem is that when it arrive to 350-400 calls, the reply fail
because it has no memory free. Now ,i have setted the pkg memory to 12MB
but at peak it reply with error. Now i shoul increase it.
Is there another method to optimize the request of these data, to avoid
every time to increase the memory parameter?
Thanks to all
Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs.
This architectue uses a TURN server and the browser is chrome with latest updates.
The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do)..
Also, does it necessitates to have Apache ruuning https on 443 ?
Thanks in advance
Sent from Samsung Mobile
-------- Original message --------
From: Gonzalo Gasca Meza <gascagonzalo(a)gmail.com>
Date:27/01/2015 4:07 AM (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ?
Have you tried passing media directly between Browser and Kamailio with any TURN server?
Are you using latest Chrome version or FF ?
A working sample config using the following architecture:
https://github.com/spicyramen/llamato/tree/LlamatoReg
signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip udp--> *
media: sipml5 ------------------------------------------------------------------------> *
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR <rahul.ultimate(a)gmail.com> wrote:
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs <rfuchs(a)sipwise.com> wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where
the SIP server sends 183 session in progress to kamailio but after that
I can only see -
"STUN: using this candidate"
"Successful STUN binding request from .."
"SRTP output wanted, but no crypto suite was negotiated"
This is fairly strange:
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated
Are you running a very old OpenSSL version by any chance?
cheers
_______________________________________________
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--
Warm Regds.
MathuRahul
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Hi Team,
We are seeing some errors in our kamilio for TCP max conn (ERROR) : 2048
(the default).
We are thinking to double the TCP connection for our kamailio registrar
server.
tcp_max_connections=4096
Is there any performance issue if we double the tcp_max_connections ?.
Currently we are setting these parameters for TCP.
tcp_connection_lifetime=3605
tcp_accept_no_cl=yes
tcp_rd_buf_size=16384
Do we need to tune any other variable if we are setting max tcp connections
to 4096 for better performance ?
Thanks for looking into this.
Regards
Varghese Paul