Hi,
I have a kamailio 4.1 as a gateway and registrar behind asterisk servers. On asterisk, I save the IP address of the originating call parsing the SIPURI or SIPCHANNEL(recvip). In both cases, I am receiving the private ip of the user agent registered with kamailio. Do I need to change anything in kamailio to receive the actual IP address. Please suggest.
Regards
Cibin
Hello all,
During authentication, is there any way to affect the password user is
sending? I do suspect not as it is a clear security matter, but won't hurt
to ask. I use auth_db module with calculate_ha1 parameter set to 1. For
reasons in integrating Kamailio into my system architecture there is a need
to store a password in some other format than for example
md5('555:domain.com:password)') while not allowing any passwords to be
stored as plaintext.
For example: md5('555:domain.com:md5('password')') but this would require
me to hash the password before authentication, in Kamailio script as I
can't do it in the clients.
Reason for this question is to have my users in a separate database, and
these users could have 0-n sip peers assigned to them, and have users
authenticate to my software and the sip peers using the same password.
cheers,
Olli
Hi,
I'm using kamailio 4.2 some of my clients are getting 401 unauthorized
after a couple of days of usage without any problems, their usernames and
passwords are still in the database server unchanged.
Any idea what's going on?
Thanks
Ahmed
Another year getting slowly to the end, an excellent opportunity to
thank to all old and new friends of Kamailio project, from contributing
code to using it, helping on mailing lists and other places online, or
advertising it across the world!
Merry Christmas and Happy Winter Holidays!
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not
getting audio back to my browser. From a packet capture I can see media
from the browser to rtpengine, and then bi-directional RTP back and forth
from my asterisk server, but rtpengine is not sending the media on to the
browser, i.e.:
browser ---------> kamailio/rtpengine <---------> asterisk
This is the output from rtpengine:
https://gist.github.com/marcantonio/bfe72644306b205cc7e1
Thanks.
Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db
instanse for dispatcher and at attrs column for our backend servers set
WEIGHT=40 and 60 for first and second server, but packets sended only at
first server ignoring weight
|---------+----------------------------->
| | dong-ming.li(a)mail.|
| | foxconn.com |
| | |
| | 2014/12/24 上午 |
| | 11:54 |
|---------+----------------------------->
>--------------------------------------------------------------------------------------------------------------------------------------------------|
| |
| To: alfred.lu(a)foxconn.com |
| cc: |
| Subject: Mediaproxy do not work well with Kamailio SIP server issue |
>--------------------------------------------------------------------------------------------------------------------------------------------------|
Secret Level: |------------------------|
| ( ) 1.Non-Confidential |
| ( ) 2.Confidential |
| ( ) 3.Secret |
| ( ) 4.Top Secret |
|------------------------|
Priority Level: |--------------------|
| ( ) 1.Normal |
| ( ) 2.Urgent |
| ( ) 3.Extra Urgent |
|--------------------|
-------------------------------------------------------------------------
Hi, sir,
We try to setup Mediaproxy on CentOS 6.6 with kamailio-4.2.1 SIP server. We
use Linphone android version as SIP client to test.
attached kamailio.cfg.txt is our kamailio-4.2.1 SIP server config file. pls
help to rename kamailio.cfg.txt to kamailio.cfg.
kamailio-4.2.1 SIP server is run on public IP server,and Linphone SIP
client is run behind NAT.
The problem is that only callee can hear caller's rtp audio data. caller
can not hear callee's audio data. And the call will auto disconnect after
about 30s time out.
Below's our modification regarding the kamailio.cfg file,pls help to check
if this is OK or not ?
#foxconn gallice add start
#### MediaProxy module
loadmodule "mediaproxy.so"
modparam("mediaproxy", "disable", 0)
modparam("mediaproxy", "ice_candidate", "high-priority")
modparam("mediaproxy", "mediaproxy_socket",
"/var/run/mediaproxy/dispatcher.sock")
modparam("mediaproxy", "mediaproxy_timeout", 500)
modparam("mediaproxy", "signaling_ip_avp", "$avp(nat_ip)")
modparam("mediaproxy", "media_relay_avp", "$avp(media_relay)")
#foxconn gallice add end
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
use_media_proxy(); #foxconn gallice add
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
{ #foxconn gallice add
#route(NATMANAGE); #foxconn gallice delete
use_media_proxy(); #foxconn gallice add
} #foxconn gallice add
}
For mediaproxy,we can see the media related processes are already run.
31212 ? SL 0:01 python ./media-dispatcher start
31218 ? SL 0:16 python ./media-relay start
*****************************************************
dong-ming / 李東明
Foxconn Electronics Inc.
HH Precision Ind. Co., Ltd.
CNSBG NSDI TIPBU R&D Dept.
LH Office
地址:深圳市龍華新區龍華東環二路二號富士康科技集團
分機 : +86-0755-2812-9588 #26109
Mobile : 13510123670
Mail : dong-ming.li(a)mail.foxconn.com,
*****************************************************
溫馨提醒﹕入廠人員須遵守富士康出入管制,請勿攜帶手提電腦、各種攝像、U盤、MP3
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This e-mail message together with any attachments thereto (if any) is
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mail from ip-->192.168.2.2
mail from pc-->MICROSOF-700F48
Version: Super Notes 1.6.4.5B
(See attached file: kamailio.cfg.txt)
Hi all,
I have tried several configure TLS in kamailio but no luck.
Please give me some suggestion that I can make it work correctly.
This is my configure in TLS module.
modparam("tls", "tls_method", "SSLv23")
modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem")
modparam("tls", "certificate",
"/usr/local/etc/kamailio/ca/kamailio1_cert.pem")
modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem")
modparam("tls", "verify_certificate", 1)
modparam("tls", "require_certificate", 1)
I am only getting issue with verify_certifiate = 1, it i let it to 0, my
client can register correctly.
When I set this flag, i got message from server as:
Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: <core>
[tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading
Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls
[tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL
routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate
Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: <core>
[tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading
I Cant add any pem file into client, i have used Blink phone but no luck.
Thank all in advance.
ThanhTruong.
I hope the subject line says it all.
I need to set up a backend to establish calls between browser clients
and usual SIP clients, and the abovementioned applications are of the
highest interest. (Of course, calls between same UAs must work, too.)
If Linphone is not going to work due to its internal issues, then
other established open source mobile apps for iOS and Android will be
fine.
Currently I have configured latest kamailio+rtpengine with configs
from here https://github.com/caruizdiaz/kamailio-ws
The calls pass this way currently:
jssip -> android: no sound from phone to the browser, i see that jssip
sends "sendonly" attribute for audio in INVITE's SDP. Audio from
browser to phone, and both video streams appear immediately,
everything is fine with them.
android -> jssip: video from browser to the phone appears in 1-2
_minutes_ after the call is answered. All other media streams are
fine.
The above results are the same in such browsers
[IP-] [ ] www-client/firefox-bin-31.3.0:0
[IP-] [ ~] www-client/google-chrome-unstable-41.0.2243.0_p1:0
With sipml currently i have no stable results, so it's hard to
describe what happens.
Please contact me if you have configs to make the needed things work,
or if you have experience of such things working stable, and can
configure it quickly.
--
Andrey Utkin