Hello,
There is standards-based support for the notion that the "identity" of the
source of an INVITE request is equal to the value of the From URI, much as the
To header indicates the AOR to register in a REGISTER request.
However, a digest challenge realm can be whatever you like. The choice
Daniel-Constantin Mierla :
> quickly checking the code it seems that the dst_ip is holding the local
> address always, even for outbound connection.
>
> Not being the author of the code, my guess is that the lookup of
> connection is done always on src_ip matching the remote peer address.
> Also,
Hi Daniel,
I 've put this patch on Kamailio 4.2.5 and it worked. I 'll reporting if have
any problem.
Thank for your patch.
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Dao
Hai Dang
Sent: Wednesday, June 03, 2015 8:58 AM
To: mico...@gmail.com; Kamailio (SER) - Use
I have a questions in regards to SIP Auth, and specifically how Asterisk 13
with PJSIP appears to send Invites.
It seems that the typical Kamailio config sends the SIP auth challenge with
the from domain as the auth realm.
however in the case of the Asterisk 13 ( PJSIP ) invite, the from domain is
Hello,
quickly checking the code it seems that the dst_ip is holding the local
address always, even for outbound connection.
Not being the author of the code, my guess is that the lookup of
connection is done always on src_ip matching the remote peer address.
Also, the structure used there is the
Hi, I noticed thath command core.tcp_list in kamcmd gives output that
does not make sense to me:
---
kamcmd> core.tcp_list
{
id: 1
type: TLS
state: CONN_OK
timeout: 3599
ref_count: 1
src_ip: 192.168.47.132
src_port: 5061
dst_ip: 10.10.
Hi Everyone it got fixed, by adjusting in kamailio.cfg and in Asterisk
dialplan. Let me keep trying with dispatch module.
Thanks.
Thank you with regards,
Gopalkrishnan N.
Mob: +91 99404 91346
VoIP call - sip:sai...@gtalk2voip.com
On Wed, Jun 3, 2015 at 12:28 PM, Daniel-Constantin Mierla wro
On 06/03/2015 12:10 AM, Mir Jee wrote:
> Hi!
> Please help me to configure rtpproxy on centos + kamailio.
>
What type of trouble are you having?
I have a tutorial written for installing an older version of Kamailio
and RTPproxy on CentOS, but this is easily changed to the current version...
ht
Hi Alex,
In contact header I'm getting last gateway is listed first , then gateways are
ordered correctly.
Let's say I have 3 gateways ordered as 1,2,3 in LCR_rule_target table
I'm getting in the contact header : 3,1,2,3
I'm using below code
if (!load_gws(1, $rU, $fu))
2015-06-02 23:16 GMT+03:00 Federico Cabiddu :
> Hi,
> to achieve the described behavior you could call t_on_branch(BRANCH_NAME)
> before calling t_relay.
> In this way each branch, after forking and before being relayed, will
> traverse the branch_route named BRANCH_NAME.
>
> http://www.kamailio.or
Ali,
That's not going to work, for reasons related to the state of the message body
at the time you're manipulating it. You need to get to the bottom of the
fundamental issue: why are the gateways not in the order you want? And you need
to fix it on that level.
--
Alex Balashov | Principal |
Hi Alex,
This is related to post Kamailio LCR Multiple choices I created before, where
last gateway is listed at the first in contact header.
So as I didn’t get a solution from your side regarding this issue , I tried to
strip first gateway from contact header.
Regards,
Ali
From: sr
Ali,With the exception of a few, pseudovariables that expose parts of a SIP message (such as $ct) are read-only, so you cannot assign values to them.Why are you manipulating the Contact header? This is not somethi
Hello,
I'm trying to change contact header in SIP response message by stripping
first characters (for testing only) as below :
$ct = $(ct{s.strip,25});
But I got below error saying that $ct is read-only:
0(15410) : [cfg.y:3419]: yyerror_at(): parse error in config file
/usr/local/etc/kam
Hi,
Any help regarding the below ?
Thanks,
Ali
-Original Message-
From: Ali Taher [mailto:ata...@vanrise.com]
Sent: Monday, June 01, 2015 3:21 PM
To: 'Kamailio (SER) - Users Mailing List'
Cc: 'Ali Taher~Vanrise Technical Support'
Subject: RE: [SR-Users] Kamailio LCR Multiple choices
Hi,
Hello.
We have been using the msilo module to handle offline SIP messaging. I have
noticed that the m_store function ends up storing all the records with the
default message id which is zero. However m_dump uses this id to figure
which messages have already been sent. Thus on calling m_dump on
regi
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