HTH
-G
On Fri, Oct 17, 2014 at 5:10 PM, Kamrul Khan wrote:
ok. I have made some changes. rather than getting the TLS configaration from a
file I added this lines:
#!ifdef WITH_TLS# - tls params -
modparam("tls", "private_key",
"/etc/certs/sip.192.168.1
HTH
-G
On Fri, Oct 17, 2014 at 5:10 PM, Kamrul Khan wrote:
ok. I have made some changes. rather than getting the TLS configaration from a
file I added this lines:
#!ifdef WITH_TLS# - tls params -
modparam("tls", "private_key",
"/etc/certs/sip.192.168.1
Hi,
I have a wired problem in my Kamailio. Over WSS, Kamailio is
dropping the websocket connection right after it is getting the BYE from
other side. So the client side is not getting any BYE. Over normal http
it is working fine. Any idea why it might happen or where to look at?
Thanks in adv
Hi,
In my Kamailio configuration I added the below lines:
route{
.
.
rtpengine_manage("replace-session-connection symmetric ICE=remove
media-address=172.16.0.150");
.
.
}
onsend_route {
if(to_ip == My.Carrier.I.P){
xlog("going to carrier");
rtpengine_offe
give me some clue what I can do? Or if more information is needed please let
know so that I can provide.
Sent from my Samsung Galaxy smartphone.
Original message
From: Kamrul Khan
Date:10-21-2014 17:45 (GMT-06:00)
To: sr-users@lists.sip-router.org
Cc:
e. Can someone please
give me some clue what I can do? Or if more information is needed please let
know so that I can provide.
Sent from my Samsung Galaxy smartphone.
---- Original message
From: Kamrul Khan
Date:10-21-2014 17:45 (GMT-06:00)
To: sr-users@lists
Hi,
We have a setup with sipml5 to kamailio. It works perfectly within local
network. In public network the signaling establishes perfectly, but most of the
time we hear no voice, sometimes we hear one sided voice and in rare cases we
hear voice from both sides. To fix this issue we configured
picyramen/llamato/blob/LlamatoReg/tls.cfg
openssl s_client -connect 172.31.27.85:5063 where this i my internal IP address
I get presented the certificates.
HTH
-G
On Fri, Oct 17, 2014 at 5:10 PM, Kamrul Khan wrote:
ok. I have made some changes. rather than getting the TLS configarati
Hi, Is there a way to know or cache the last destination selected from
dispatcher list?
Thanks in advanced!
Date: Mon, 20 Oct 2014 14:52:38 -0400
From: vipki...@gmail.com
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Hash Table capacity
Alex,Yes, linked-lists attached to a hash
buck
, 17 Oct 2014 19:56:50 -0300
From: b...@langfeld.co.uk
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Configuring TLS and WSS with Kamailio
Out of curiosity, why don't you use the pre-built packaged binaries instead of
building from source?
On 17 October 2014 19:53, Kamrul Khan wro
S and WSS with Kamailio
Out of curiosity, why don't you use the pre-built packaged binaries instead of
building from source?
On 17 October 2014 19:53, Kamrul Khan wrote:
Hi,
Im trying to connect my WebRTC clietn
to kamailio via WSS. I successfully connected it via WS but having
trouble
Hi,
Im trying to connect my WebRTC clietn
to kamailio via WSS. I successfully connected it via WS but having
trouble connecting it via WSS. My first issue is I get error messages
while i try to compile TLS module(console log in the end of the
document). But, still it creates tls.so file. So i co
Hi,
Im trying to compile TLS module in my kamailio. Below is my console log. It
came up with lots of warnings and errors. But it generates the tls.so. What im
supposed to do now and how to fix it?
xxx@ubuntu:/usr/local/src/kamailio-4.1/kamailio$ sudo make -C modules/tls
make: Entering directory
hi,
thanks for your reply. so what you are saying is active calls will continue
with the same server ? for example, lets say i have two entries in my
dispatcher.
10.1.1.110.1.1.2
There are already 10 active calls between my Kamailio and 10.1.1.1 and i just
removed 10.1.1.1 from dispatcher. So no
htable module about how to define origin hash table.
Cheers,
Daniel
On 28/08/14 01:16, Kamrul Khan wrote:
Hi,
I want to append the origin header of my incoming
websockets to the SIP INVITES. I tried something l
Hi,
I want to append the origin header of my incoming websockets to the SIP
INVITES. I tried something like the below but didnt work:
if (ws_handle_handshake()) {
if(method=="INVITE"){ insert_hf("Origin:
$hdr(Origin)");
Hi,
We have a Kamalio a configuration that forwards calls to a freeswitch server
(adding record-route). The SIP signaling becomes successful till Kamailio
receives 200 OK with session description from our carrier. After kamailio
forwards the 200 OK with session description to Freeswitch, Kamail
#x27;.
Let me know if works.
Cheers,
Daniel
On 19/08/14 17:20, Kamrul Khan wrote:
Hi Daniel,
Thanks for your reply. I need username, uri and response.
We will also pass some custom headers with our SIP
ok at it once we get a chance.
Cheers,
Daniel
On 18/08/14 23:58, Kamrul Khan wrote:
I want to send the authorization information
that I received from my SIP devices to another serve
xhttp:request
Hello,
can you check to be sure that event_route is not in a disable part
of config by #!ifdef conditions?
Cheers,
Daniel
On 16/08/14 02:33, Kamrul Khan wrote:
i am trying to connect to my kamailio
I want to send the authorization information that I received from my SIP
devices to another server for authentication. How to get the values of
Authorization header. I tried using hf_value and hf_value2 but for
@hf_value2.Authorization.username I get "Digest" as output. Is there any other
w
@lists.sip-router.org
Subject: Re: [SR-Users] event_route xhttp:request
Hello,
can you check to be sure that event_route is not in a disable part
of config by #!ifdef conditions?
Cheers,
Daniel
On 16/08/14 02:33, Kamrul Khan wrote
i am trying to connect to my kamailio through websocket. I added the below code
in my config file:
event_route[xhttp:request] { xlog("websocket connection request");
set_reply_close(); set_reply_no_connect(); if (ws_handle_handshake())
{ exit;
Hi, Im new to kamailio. Can someone tell me where to find the logs (from xlog()
or log() ) ? thanks in advanced ___
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