Hi,
just remove:
#!define WITH_ASTERISK
From your kamailio.cfg and restart it.
--
Daniel Grotti
On 01/02/2017 06:36 PM, Manoj Gupta wrote:
Hi Daniel,
If it is not too much asking, please Can you help me in giving instructions on
how to disable kamailio from my asterisk. I am much
Kamailio-asterisk doc:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
There are tones of documentation about kamailio out there.
Consider to check SIP protocol as well (rfc3261).
--
Daniel Grotti
On 01/02/2017 06:18 PM, Manoj Gupta wrote:
Now we are getting
You should add "ims.airtel.in" as kamailio local domain, in your
kamailio.domain table.
--
Daniel Grotti
On 01/02/2017 05:31 PM, Manoj Gupta wrote:
HI Daniel,
Here is another problem. Kamailio responds with SIP/2.0 478 Unresolvable destination
(478/SL) to REGISTER req
is a standard log line\n");
After restarting the processes, what do you see in /var/log/kamailio ?
What do you get from kamctl fifo debug ?
--
Daniel Grotti
On 01/02/2017 04:13 PM, Manoj Gupta wrote:
Hi Daniel,
Thanks for the quick revert!
Yes we have configured the same in /etc/rsyslo
Hi,
have you configured kamailio in order to log to /var/log/kamailio
instead of syslog ?
https://www.kamailio.org/dokuwiki/doku.php/utils:basic-syslog-configuration
--
Daniel Grotti
On 01/02/2017 03:36 PM, Manoj Gupta wrote:
Request to all – Please help we are BADLY stuck
Hi,
not sure if I understood it right but, have you defined the
advertised_address ? That should be used in Via and RR as well:
https://www.kamailio.org/wiki/cookbooks/4.4.x/core#advertised_address
Daniel
On 12/29/2016 12:09 AM, Pranathi Venkatayogi wrote:
I implemented full NAT logic as
Yeah,It was auth_db, not auth.Good night! :PDaniel G.___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Jon,
isn't it "invalid user" ?
> auth_db [authorize.c:199]: get_ha1(): no result for user '48481628780@'
http://www.kamailio.org/docs/modules/4.4.x/modules/auth_db.html
Daniel
On 12/19/2016 03:23 PM, Jon Bonilla (Manwe) wrote:
Hi all
I'm getting an auth challenge result of -3 and I
Hi,
you can set TOS value:
http://www.kamailio.org/wiki/cookbooks/4.4.x/core#tos
just map your DSCP value into the TOS value.
--
Daniel
On 09/08/2016 10:32 PM, Marcelo Bonin wrote:
I need to set DSCP values in Kamailio, I couldn't found a way to set in
the configuration.
Any help will be
Hi
this shoud work:
Fetch the current debug level:
# kamctl fifo debug
Set level to 2:
# kamctl fifo debug 2
Cheers,
Daniel
n 08/24/2016 04:54 PM, Federico San Martín wrote:
Hi guys, sorry for the newbie question, I searched for it and couldn't
find any reference, is there a way to
t;
Sent: Tue, 16 Aug 2016 18:26:55 +0200 (CEST)
Subject: Re: [SR-Users] Stress Testing
Hi Dan,
We have checked all of that and we also not using pike its really strange
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 16 Augus
Hi,
a couple of suggestions:
you may running out of RTP ports.
kamailio may start to block requests if you are using pike module.
UAC may start to de-register after a short expires time, so they won't
be reachable anymore if they do not refresh their registration.
You may check your natping
-Route:
<sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>.
Content-Type: application/sdp.
Content-Length: 249.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4.
s=SIP Call.
c=IN IP4 x.x.x.x.
t=0 0.
m=audio 18838 RTP/AVP 3 101.
c=IN IP4 83.147.65.
Ciao Laura,
would be interesting to see the INVITE from kamailo2-->Cisco and see the
headers there, as well as the 180/200 from Cisco->kamailio2.
As Carsten said, probably Cisco is messing up From/To headers. The 9990
color is not present in any of the INVITEs you provided, so would be
nice to
Hi,
Did you forget to fix contact and sdp in your onreply_route?
Daniel
On Jun 5, 2015 3:30 AM, Alex neiroma...@gmail.com wrote:
Hello!
Please help to fix problem with sdp headers
UAC Inet - (X.X.X.X) Kamailio (192.168.30.250) - Asterisk (192.168.30.2)
When i call from UAC to 9002 i
Hi,
if the app crashes, it will just de-register the user after the
expiration time, since no re-register came in.
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 04/20/2015 10:34 AM, Filip Malenka wrote:
Thank you
Hi,
here you are:
https://tools.ietf.org/html/rfc3665#section-2.4
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 04/20/2015 10:10 AM, Mickael Marrache wrote:
Hi,
You need to configure your client to un register when
Hi,
you may want to use:
http://www.kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.p.keepalive_timeout
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 04/20/2015 10:34 AM, Filip Malenka wrote:
Thank you
Hi,
have you read this ?
http://www.kamailio.org/docs/modules/4.2.x/modules/mohqueue.html
Otherwise you would need a B2BUA (like Asterisk, Freeswitch, Sems) in
order to provide moh functionality.
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria
flow ?
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 04/16/2015 07:23 PM, Dmitry Sytchev wrote:
Hi all!
I can't make my Kamailio to correctly relay stateful ACKs for 200
OK. It takes URI from Route: header and trying relay
Hi,
please also check if you have set aliases (kamailio.cfg global
parameters) to the IP address in the R-URI.
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 04/16/2015 07:23 PM, Dmitry Sytchev wrote:
Hi all!
I can't make
Hi,
q value is used in location table for serial forking.
If you have a subscriber with 2 contact registered you can use the q value to
load one contact first (the one with the higher q) and afterwards load and call
the second one.
Highest is the value stored, highest is the priority.
Daniel
Hi,
a call trace could help you to understand why your server is not
receiving response (which one actually ?) form the Cisco.
Is that because the Cisco didn't receive your SIP message? Or is it
because Cisco replied but the response didn't reach your server ?
Try to make a sip trace in order to
On 06/24/2014 12:18 PM, Jack Smith wrote:
Dear friends,
Just new to this mailing list, new to Kamailio, and new to running a
sip server.
I would like to try only a redirect server. I had a look at
http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg
which was
Hi,
why you don't user functions from rptproxy module ?
http://kamailio.org/docs/modules/4.1.x/modules/rtpproxy.html#idp79896
Daniel
On 04/11/2014 03:02 PM, Daniel Ciprus wrote:
If you're going to recommend to upgrade/or use something else - this is
not feasible for some weird reasons. Is
Rome!
Great choice! :)
Daniel
On 04/08/2014 10:31 AM, Carlos Ruiz Díaz wrote:
Hi Carsten,
that's right, I'm replying to this while I wait for a tour to the
Vatican City :-D ;-)
Cheers,
On Tue, Apr 8, 2014 at 3:27 AM, Carsten Bock cars...@ng-voice.com
mailto:cars...@ng-voice.com
Hi Ravi,
if you media/rtp proxy is receiving such packet loss, it means that
something behind him is cutting the traffic off, somehow.
So you should investigate there, in their configuration.
If your analysis is right, it seems your loosing around 80% of your packets.
Do you have any router,
Hi,
When you don't use RTPproxy, RTP traffic is sent end-to-end.
What is the RTP path in case of client-to-client RTP ?
Is it the same ?
Daniel
On 02/25/2014 06:34 PM, Ravi wrote:
Dear Daniel,
Thank you for the response,
Do you have any router, firewall something between clients and
Hi,
Yes packet loss is very high.
Please investigate around that.
Daniel
Ravi wingsravi...@gmail.com wrote:
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
Hi Ravi,
yes it means that when RTP traffic passes through your media-relay you have
traffic, if you don't use media-realy RTP traffic is end-to-end between clients.
To check jitter and other values you can capture your SIP/RTP traffic on your
kamailio server with tcpdump for example and
routing logic
route{
if ($ua==friendly-scanner) {
sl_send_reply(200,OK);
exit;
}
On Nov 26, 2013, at 5:29 PM, Daniel Grotti dgro...@sipwise.com
mailto:dgro...@sipwise.com wrote:
Hi,
you can check the User-Agent reference $ua
Hi,
you can check the User-Agent reference $ua, if it is equal to
friendly-scanner, just send back a reply with sl_send_reply(200, OK)
Daniel
On 11/26/2013 10:53 PM, Joli Martinez wrote:
How can I do this? Is there an article I can reference or something? I am
new to kamailio and not sure
Hi,
are you handling method MESSAGE in your kamailio.cfg ?
if(is_method(MESSAGE))
{
do something
}
Daniel
On 11/22/2013 03:10 PM, Karthikeyan R wrote:
Is Kamailio supports SIP MESSAGE request? Please let me know how to
configure.
Regards,
Karthikeyan R
DISCLAIMER:
Hi,
also you can try to use the IPOPS module, for example:
if (is_in_subnet(10.0.123.123, 10.0.123.1/24)) {
xlog(L_INFO, it's in the subnet\n);
xlog(L_INFO, Skip NAT test\n);
...
...
}
http://kamailio.org/docs/modules/4.0.x/modules/ipops.html#idp42416
Daniel
On 11/13/2013 11:42
Rori,
this is not the way.
Please subscribe to the mailinglist here:
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Daniel
On Thursday, November 14, 2013 22:11 CET, Roni | r...@pomvan.co.uk wrote:
may i join
___
SIP
Hi,
have a look at Kamailio transformation:
http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#uri_transformations
Daniel
On 11/13/2013 12:19 PM, Daniel-Constantin Mierla wrote:
Hello,
try:
$(ru{uri.param,user}{s.substr,3,0})
Cheers,
Daniel
On 11/13/13 12:13 PM, Oliver
Hi Joli,
if you need to use an external media-relay, you need to change the IP in
the SDP for INVITE (main route) and 200 OK (onreply route):
You can try to use the fix_nated_sdp(2, EXT_IP_ADDR).
http://kamailio.org/docs/modules/4.0.x/modules/nathelper.html#idp15360968
make sure that
Hi,
also you can choose the outbound socket to use by setting '$fs'
pseudovariable or use force_send_socket() function.
http://www.kamailio.org/wiki/cookbooks/4.0.x/core#script_operations
http://www.kamailio.org/wiki/cookbooks/4.0.x/core#force_send_socket
Daniel
On 11/12/2013 08:57 AM,
.
Daniel
On Thursday, October 31, 2013 22:10 CET, Lucas Girard lgir...@commpartner.net
wrote:
Yes, I have used the r flag, in rtpproxy_manage(). I have not offer or
answer functions, just manage.
On Thu, 2013-10-31 at 21:41 +0100, Daniel Grotti wrote:
Hi Lucas,
are you using
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.pulla...@plintron.com wrote:
Hi all,
I have a requirement , i.e. I need to capture 200
Hi Girard,
looks like the Provider are sending RTP from port 5392 instead of 5394
as in SDP, rtpproxy send traffic to 5392 as well.
Daniel
On 10/31/2013 04:13 PM, Lucas Girard wrote:
Hi all,
I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is used.
I have the system
about which ports should be used.
On Thu, 2013-10-31 at 16:18 +0100, Daniel Grotti wrote:
Hi Girard,
looks like the Provider are sending RTP from port 5392 instead of 5394
as in SDP, rtpproxy send traffic to 5392 as well.
Daniel
On 10/31/2013 04:13 PM, Lucas Girard
Hi all,
I would like to use kamailio 3.1 with TLS and verified also a client
certificate.
My tls.cfg file is as follow:
--- tls.cfg
.
.
[server:MY_IP:5061]
method = TLSv1
verify_certificate = yes
require_certificate = yes
private_key = default_key.pem
certificate =
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