Hi Team,
I am using Jitsi client with Kamailio. I was able to IM between the clients. How can I control the Message routing in Kamailio script. Please help me with any link.
Regards,
Surya.
Hi all.
I have this setup.
Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout
FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks.
AJ
Hi.
Asterisk users are fine.
Register subscribers Kamailio is my problem.
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sipusers", "1")) { ##### OK Asterisk Users are no problem
#!else ##### But if the user is no asterisk , simply refuse registration, without Respect the jump to validate users Kamailio
if (!auth_check("$fd", "subscriber", "1")) {
#!endif
#####
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: auth_db [authorize.c:498]: auth_check(): realm [192.168.65.132] table [sipusers] flags [1]
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm '192.168.65.132' not found
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: auth_db [authorize.c:264]: digest_authenticate_hdr(): no credentials
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: auth [challenge.c:127]: get_challenge_hf(): build_challenge_hf: realm='192.168.65.132'
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth: 'WWW-Authenticate: Digest realm="192.168.65.132", nonce="VUDgjFVA32CFnv6lYnT/fxOzv3M7Mw7n"#015#012'
Thanks.
De: sr-users [mailto:sr-users-bounces@lists.sip-router.org] En nombre de SamyGo
Enviado el: jueves, 30 de abril de 2015 4:39
Para: Kamailio (SER) - Users Mailing List
Asunto: Re: [SR-Users] [SR_USers] Authenticate asterisk-kamailio
Hi,
The important thing to consider here is this line.
#!define WITH_ASTERISK
so if you've defined this on the very top of your kamailio.cfg then it will go and check username/passwords from the sipusers table from the Database defined by this: DBASTURL
if (!auth_check("$fd", "sipusers", "1")) {
Make sure you've the same user defined and username and passwords are defined in these columns:
modparam("auth_db", "user_column", "name")
modparam("auth_db", "password_column", "sippasswd")
the sipasswd column is text based so dont use encrypted PASSWORD() in there.
Once you follow these steps your user should get registered.
Hi,
In our config we don't use presence, but I do have a question about
transactions.
We're using dialogs as well as transactions in the routing. To keep
things simplyfied, let's assume we have the following stripped out route
setup:
request_route {
if (is_method("CANCEL")) {
if (t_check_trans()) {
t_relay();
}
exit;
}
# As stated in documentation, this call could stop further
# routing from here on out
t_check_trans();
if (is_method("INVITE")) {
if (ds_is_from_list()) {
setflag(DLG_FLAG);
}
setflag(FLT_ACC);
}
if (lookup("loction") == -1 ) {
if(load_gws(1)) {
t_newtran();
t_on_failure("TFAIL");
}
}
}
route[TFAIL] {
xlog("L_ERR", "Failed setting up transaction");
}
What I'd like to know is when or even if we need to be releasing the
transaction at some point. From what I understand in the presence
example I should be releasing a transaction when I use t_newtran, but I
haven't found any evidence of needing it when working with t_relay or
t_check_trans.
For example, could it be needed to do a t_release() on a
dialog_terminated or dialog_end event?
In the above example, would it just be enough to add t_release() to the
failure route TFAIL?
Further details about how and/or when the transactions are cleaned up
internally would be greatly appreciated!
--
Kind regards,
Dirk Teurlings
I have a general question maybe somebody can help me out with. We have
a new SIP Trunk setup with a provider. The SIP Trunk has a username of
'jane' and it handles 400 DIDs. When the incoming INVITE from the
provider comes in, the URI in the Invite is the username of the trunk
while the To header actually has the dialed number. As long as I can
remember I have always seen the Invite as having the same URI as the To
Header in the initial INVITE but not so here.
18:17:34.786 SIP.STACK MSG INVITE
sip:jane@10.5.5.5:5060;transport=UDP SIP/2.0
18:17:34.787 SIP.STACK MSG Via: SIP/2.0/UDP
10.10.10.10:5060;branch=z9hG4bK290hka10e8j11dn7k0k0.1
18:17:34.788 SIP.STACK MSG From: "Test
"<sip:9541112222@voip.xxx.net;user=phone>;tag=1574552918-1430345854933-
18:17:34.788 SIP.STACK MSG To:
"Customer"<sip:2222222222@voip.xxx.net>
I am inclined to believe this is perfectly normal and compliant but let
me know what you think.
Thanks,
--
Technical Support
http://www.cellroute.net
Hi
I've realised that I should put '^' after "Dlg=>" (rather than before),
i.e. sht_rm_name_re("Dlg=>^$var(callid)::tenant"); - this works.
But it doesn't seem to like '$' at the end of the regexp - Kamailio failed
to start up with error:
"ERROR: htable [ht_var.c:176]: pv_parse_ht_name(): wrong format". Any way
to use the end-of-line symbol?
Cheers,
Yufei
> ------------------------------
>
> Message: 10
> Date: Wed, 29 Apr 2015 18:01:26 +0100
> From: Yufei Tao <yufei.tao(a)gmail.com>
> To: sr-users(a)lists.sip-router.org
> Subject: [SR-Users] sht_rm_name_re() question
> Message-ID:
> <
> CAJwP0iTwhcTj6SJWABHj+eGDqQv13wV_e1wC2s1Na2+WoSuJGA(a)mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi
>
> I've hit a problem with sht_rm_name_re() in htable module. I was calling it
> like this:
>
> sht_rm_name_re("Dlg=>$var(callid)::tenant");
>
> But when I used sipp to generate 600 concurrent calls for example, I called
> this function when receiving BYE. But it removed more entries than it
> should. Seems when removing removing some entry with callid
> 1-3798(a)192.168.60.80, it also removed entries for [1-9]
> 1-3798(a)192.168.60.80.
>
>
> How do I add '^' in front and '$' at the end of the regexp string when
> calling this function please? Here I don't really need regexp in fact, just
> want to do an exact match. But I didn't find any other functions for
> deleting entries.
>
> Thank you!
> Yufei
>
Hello,
I'm new with Kamailio and have to do a research project with Kamailio server. My task is to realize a SIP infrastructure with NATing and a High Availability Cluster with two Kamailio server.
For the NAT problem I found a solution with SEMS, but for security I need authentication on the SBC. I have no idea to synchronize the user authentication information from Kamailio to SEMS. Perhaps there is a more useful solution available? The SIP server is behind the NAT in a private network and the SBC to be placed in the DMZ.
A new feature of Kamailio 4.2 is a replication for High Availability (HA), but I did not found any information about this.
Is there somewhere a documentation or howto where I can look?
I need only short keywords and not a complete description.
Many thanks,
Kai
hello
i have configured msilo module successful but there isn't any voice with
msilo. i debug and compare the call flow with and without msilo. with msilo
in call flow it didn't follow to relay and consequently didnt activate
rtpproxy befor the call is answered. due to which during call there isn't
any voice. please help
--
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