Front of my eyes and I did not see it.
Thank you very much.
-Mensaje original-
De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] En nombre de
Daniel Tryba
Enviado el: jueves, 30 de abril de 2015 13:44
Para: sr-users@lists.sip-router.org
Asunto: Re: [SR-Users] [SR_USers] Auth
On Thursday 30 April 2015 13:23:25 Mauricio Tejeda wrote:
> #!ifdef WITH_ASTERISK
>
>if (!auth_check("$fd", "sipusers", "1"))
> { # OK Asterisk Users are no problem
>
> #!else
>
Hi.
Asterisk users are fine.
Register subscribers Kamailio is my problem.
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sipusers", "1")) {
# OK Asterisk Users are no problem
#!else
I'd like you to google around, there is a function available from another
module which will apply the changes in SIP Message.
On Thu, Apr 30, 2015 at 9:51 AM, Ali Jibran wrote:
> Perfect. Yeah got the working.
>
> Just one last issue. I don’t think this is rewriting the header. When I
> log the
Hi,
Thanks for all responses, I think we can do completely without the
t_newtran too now. We got it originally from one of the older CNXCC
routing examples. But it seems we can skip it entirely indeed.
TMX might also be good for us to use earlier transaction detection. I'll
look into that as
Perfect. Yeah got the working.
Just one last issue. I don’t think this is rewriting the header. When I log the
header again after the changes it still shows me the old values.
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
SamyGo
Sent: Thursday, April 30, 2015 6:50
t_on_failure("F_VOIP") to be used before t_relay();
That will arm the call to go to F_VOIP on failure responses.
On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran wrote:
>
>
> #!ifdef WITH_FREESWITCH
>
> if(is_method("INVITE") && route(FROMFREESWITCH))) {
>
> xlog("L_INFO" ,"[$
On 4/30/15 9:28 AM, Alex Balashov wrote:
No, that's not correct. The provider needs to send DNIS in the RURI in these
cases, and providers should have a setting to enable this. It does require
overriding the Contact binding of the registrant (if applicable), which is not
RFC-compliant, but tha
#!ifdef WITH_FREESWITCH
if(is_method("INVITE") && route(FROMFREESWITCH))) {
xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call from FreeSWITCH
needs to be sent TOVOIP \n");
route(TOVOIP);
t_on_failure("F_VOIP");
exit;
No, that's not correct. The provider needs to send DNIS in the RURI in these
cases, and providers should have a setting to enable this. It does require
overriding the Contact binding of the registrant (if applicable), which is not
RFC-compliant, but that's okay.
--
Alex Balashov | Principal | E
On 4/30/15 7:35 AM, Alex Balashov wrote:
On 04/30/2015 07:31 AM, Andres wrote:
I am inclined to believe this is perfectly normal and compliant but let
me know what you think.
Yep, it's normal. Moreover, only the RURI value should be used for
routing purposes or for anything else that's conse
On 04/30/2015 07:31 AM, Andres wrote:
I am inclined to believe this is perfectly normal and compliant but let
me know what you think.
Yep, it's normal. Moreover, only the RURI value should be used for
routing purposes or for anything else that's consequential in relation
to the destination;
I have a general question maybe somebody can help me out with. We have
a new SIP Trunk setup with a provider. The SIP Trunk has a username of
'jane' and it handles 400 DIDs. When the incoming INVITE from the
provider comes in, the URI in the Invite is the username of the trunk
while the To h
This happens only with one trunk. We also have plivo trunks and it works
fine.
Syslog show nothing when this message comes
Started with debug mode and saw that realms didn't mach. Sorry for stupid
questions. All works fine.
2015-04-30 13:24 GMT+03:00 Daniel-Constantin Mierla :
> Hello,
>
> doe
Hi
I've realised that I should put '^' after "Dlg=>" (rather than before),
i.e. sht_rm_name_re("Dlg=>^$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 f
Hello,
does that happen in all cases or just for some records? Can you rung
with debug=3 and check the syslog messages for what happens at that
moment when 401 is processed?
Cheers,
Daniel
On 30/04/15 11:37, Yuriy Gorlichenko wrote:
> Hello. We have an issue with REGISTER to Provider. When Provi
Hello,
there is a rather new lightweight mechanism to detect retransmissions
without creating the transaction before relaying. It is already part of
4.2 and in the default configuration file -- see t_precheck_trans() from
tmx module.
Cheers,
Daniel
On 30/04/15 10:31, Mickael Marrache wrote:
> H
Hello. We have an issue with REGISTER to Provider. When Provider answers
401 Kamailio don't send any REGISTER with digest auth
IP ourservice.com.5068 > provider.dev.5060: UDP, length 468
E...U...@.3'
..AREGISTER sip:provider.dev SIP/2.0
Via: SIP/2.0/UDP ourservice.com:5068
;branch=z9hG
There are really no healthy applications of Kamailio in which processing of a
request should take a long time, due to Kamailio's limited number (necessarily
so) of SIP worker threads. So, the temporal delta from initial message receipt
to t_relay() should not be high, and once t_relay() is calle
Thanks for the awesome detailed explanation :)
I talked to Voipfone(trunk) and they only allow registered endpoints to
make/receive calls. So I can't do IP Auth as of now.
I'll try the other method by rewriting $fu and $du. Hopefully that'll work.
Thanks for the help again.
AJ
> On 30-Apr-2
Hi,
I think one would like to create transaction ASAP to absorb retransmissions.
In some scenarios, processing a message can take time and retransmissions
are likely to happen, and you don't want the retransmissions to be processed
again (which can lead to errors in some scenarios). t_relay is gen
Dirk,
Why do you need to create new transactions yourself with t_newtran()? Nonexotic
applications of Kamailio usually don't require this. A transaction is
automatically created after t_relay() is called, and its lifecycle is
automatically managed - including destruction - based on the replies
Hello Kai,HA cannot be described as a concrete feature in Kamailio per se. It is rather that Kamailio has certain capabilities that lend themselves to incorporation into some user-designed HA methodology.This inc
# - msilo params -
#!ifdef WITH_MSILO
modparam("msilo", "db_url", "mysql://kamailio:abc@localhost/kamailio")
modparam("msilo", "from_address", "sip:regist...@sscc.tk")
modparam("msilo", "contact_hdr", "Contact: regist...@sscc.tk;msilo=yes\r\n")
modparam("msilo", "content_type_hdr", "Con
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 S
Could you provide your section of the code where you're using msilo ? how
you're using it !
I just wonder how could it just tell kamailio to skip rtpproxy stuff !!
On Wed, Apr 29, 2015 at 11:55 PM, sscc wrote:
> i have configured msilo module successful but there isn't any voice with
> msilo. i
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"))
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")) {
Hi
I'm still struggling with this problem. My setup is:
mobile sipproviderA kamailio1 - kamailio2 - SIP
user 2001
The SIP user calls the mobile and every thing works fine until the mobile
hangup and the sipproviderA sends a BYE to kamailio1. That bye message has
to "R
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 v
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
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