I have kamailio as a redirect server. It responds with 302 to invites.
The problem I am having is it continues to send the 302 multiple times.
How can i get it to stop sending once it gets the first ACK
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SIP Express Router (SER) and Kamailio (OpenSER)
On an 302 message caught in failure_route Is it possible to get to access
$hdr(xxx) of the 302.
When I do this $hdr(xxx) it is null.
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thanks
11.08.2016, 17:06, "Daniel Tryba" :
> On Thu, Aug 11, 2016 at 04:46:38PM +0300, jaflong jaflong wrote:
>> kamailio A relays to kamailio B
>> kamailio B send 300 redirect back to kamailio A
>>
>> In this setup how do I catch the redirect response bac
kamailio A relays to kamailio B
kamailio B send 300 redirect back to kamailio A
In this setup how do I catch the redirect response back from kamailio B
What conditions should I do to catch and further process the redirect message
Regards
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SIP E
Hi Everyone,
What is a accurate way to get current call count. I am only relaying calls.
Initialy I tried the code below. It seemed to work, it reported the correct
number of calls I had dialed. I get the result through kamcmd htable get
command.
--
un
Hi
When I receive a ACK how do I check that it belongs to a current dialog.
Is there a function that will do this
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see the ACK is related to the "200 OK". Do you see retransmissions from kamailio? Regards,Kristian HøghUni-tel A/S On Thursday 10 March 2016 04:00:24 jaflong jaflong wrote:> Hi Everyone>> I am manually processing sdp and handling media so I want to repond to ACK myself instead of
Hi Everyone
I am manually processing sdp and handling media so I want to repond to ACK
myself instead of forwarding on for media server.
I catch the method=="ACK" condition. From within this condition how do I
retrieve avp variables that were set in the INVITE.
Requesting $avp("myvar") comes ou
Hi everyone
How can I generate a random port number, one that is currently available to
bind to.
I want to assign this to the media parameter in the sdp
thanks
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hone-event/8000\r\n" +
>
> "a=ptime:20\r\n" +
>
> "a=sendrecv\r\n";
>
> set_reply_body($var(sdp), "application/sdp");
>
> t_reply("200", "OK");
>
> Regards,
>
> Kristian Høgh
>
> Uni-tel A/S
>
>
>
> "a=ptime:20\r\n" +
>
> "a=sendrecv\r\n";
>
> set_reply_body($var(sdp), "application/sdp");
>
> t_reply("200", "OK");
>
> Regards,
>
> Kristian Høgh
>
> Uni-tel A/S
>
> On Monday 07 March 2016 14:15:
back to a INVITE an 200 OK with a sdp body
07.03.2016, 09:56, "Daniel-Constantin Mierla" :
> Hello,
>
> I haven't seen this function to be exported to configuration file? In
> which module docs did you find it?
>
> Cheers,
> Daniel
>
> On 06/03/16 04:08, j
Hi
Is the function 't_reply_with_body' currently available in Kamailio. I am using
4.3.5
I get the error0(7176) ERROR: [cfg.y:3295]: yyparse(): cfg. parser:
failed to find command t_reply_with_body (params 3)
thanks
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SIP Express Router (SER)
Hi,
I am getting a 403 not relaying response.
This is the part of the config
How would I print the value of var myself, from_uri and uri
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_ur
I need some instructions on how to setup dbtext properly
I generated dbtext files through "./kamdbctl create"
I then add user through "./kamctrl add user1 user1"
It seems to have added the user because when I do the show './kamctl show
user1' command it displays
[1, ' user1', '10.1.1.1', ' u
enge.c:127]: get_challenge_hf():
build_challenge_hf: realm='10.1.1.1'
1(11480) DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth:
'WWW-Authenticate: Digest realm="10.1.1.1",
nonce="VVn9M1VZ/AcGIj0QNJ3dGrPP3Zs73rle"
still user cannot be found
18.05.2015,
cat > /usr/local/etc/kamailio/dbtext/subscriber
username(str) password(str) ha1(str) domain(str) ha1b(str)
user1:user1:xxx:10.1.1.1:xxx
9350) DEBUG: auth_db [authorize.c:498]: auth_check(): realm [10.31.8.1] table
[subscriber] flags [1]
2(9350) DEBUG: auth [api.c:96]: pre_auth(): auth: dige
create the tables? At least the location is not valid for
> any of the past several stable versions.
>
> Cheers,
> Daniel
>
> On 15/05/15 12:01, jaflong jaflong wrote:
>> HI
>>
>> When using dbtext setup I receive a CRITICAL error and shut down. This is
>>
HI
When using dbtext setup I receive a CRITICAL error and shut down. This is the
debug log.
My config settings are described below.
Please advise how I should fo the correct dbtext config
9(21196) DEBUG: [sr_module.c:920]: init_mod_child(): DEBUG:
init_mod_child (-4): acc
9(21196) DEBUG: [
nd test again, you should get more details
> about what kamailio tried to do and what is the potential error (maybe
> some empty line).
>
> Cheers,
> Daniel
>
> On 08/05/15 14:53, jaflong jaflong wrote:
>> db_text error
>>
>> I am getting the following
db_text error
I am getting the following error on using db_text
0(6638) INFO: db_text [dbt_base.c:99]: dbt_init(): using database at:
/usr/local/etc/kamailio/ 0(6638) ERROR: db_text [dbt_base.c:207]: dbt_query():
table version not loaded! (too few columns)
0(6638) ERROR: db_text [dbt_base.c
Hi Guys
I want to archive this setup
Kamailio to switch called number and the switch to be hidden from the client so
there should be no mention of the switched number
in any of the messages back to the client.
for example
phone client calls "alex"
kamailio to switch "alex" to "54321" and send t
On a request received over websocket how can the ip address of the http server
that jssip library connected through and send the call from be know?
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26.02.2015, 11:47, "Daniel-Constantin Mierla" :
> It seems that the value in a datetime column cannot be converted -- can
> you show the result of the same query in mysql client?
>
> Cheers,
> Daniel
>
> On 26/02/15 00:45, jaflong jaflong wrote:
>> I am ge
I am getting errors when I run the following code
if (sql_xquery("mydb", "select * from account where account = 'demo'", "res")
== 1) {
xlog("L_INFO", "my number: $xavp(res=>number)\n");
} else {
xlog("L_WARN", "Connection forbidden from $si\n");
are you using? You should install it via packages,
> the name on debian is like libmysqlclient-dev...
>
> Cheers,
> Daniel
>
> On 25/02/15 14:19, jaflong jaflong wrote:
>> On compiling I get this error
>>
>> km_dbase.c:38:25: fatal error:
On compiling I get this error
km_dbase.c:38:25: fatal error: mysql/mysql.h: No such file or directory
#include
^
I have installed mysql using the standard install from tar.gz binary
This places the prefix for install in /usr/local/mysql
the include dir is /usr/local/m
Hi Everyone,
Does anyone have a example of the config where I can get the following to work
I want Kamailio to process websocket converting wss to tcp and srtp to rtp and
forward to asterisk as tcp and rtp
incoming call on websocket (wss rtp/savpf) --> kamailio/rtpproxy (protocol
- convert
Hi list,
I am getting this error "ERROR: websocket [ws_handshake.c:153]:
ws_handle_handshake(): retrieving connection"
Any idea why?
This is the debug log
12(24113) DEBUG: [receive.c:154]: receive_msg(): After parse_msg...
12(24113) DEBUG: xhttp [xhttp_mod.c:358]: xhttp_handler(): new fake
This is webrtc, using Kamailio with websocket relay to Asterisk.
I am not using rtpproxy
07.04.2014, 22:49, "Kelvin Chua" :
> is this webrtc?are you using rtpproxy?
>
> Kelvin Chua
>
> On Mon, Apr 7, 2014 at 6:37 AM, jaflong jaflong wrote:
>> Hi,
>>
>
Hi,
I am at the point where connection is established and no apparent errors are
reported.
However audio is not output.
The rtp traffic seems to be transfering between the points as conclueded
because Asterisk debug log shows
Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq
Hi,
Is it possible to use pcmu start to end, so I send pcmu instead of opus from
the browser?
Regards
04.04.2014, 12:29, "Jon Bonilla (Manwe)" :
> El Fri, 04 Apr 2014 08:18:22 +0200
> Rainer Piper escribió:
>
>> Hallo,
>> my guess is the audio codec opus
>>
>> asterisk can NOT do transcod
Hi List,
Can anyone help me understand why this is getting rejected
Please note the specific message further dow the log.
"Failed to parse SessionDescription. Failed to parse audio codecs correctly"
This is on Chrome.
On Firefox There is a further message in the console
"Could not negotiate
protocol $proto");
xlog("L_INFO", "HTTP Request Received\n");
..
Going to https://10.1.2.3:6443 gives this
Received HTTP request to / from [10.1.1.1:58179] with protocol tls
19.03.2014, 19:50, "Olle E. Johansson" :
> On 19 Mar 2014, at 16:4
the examples directory in Git. Have you tried using this? Peter On 19 March 2014 15:50, Olle E. Johansson <o...@edvina.net> wrote: On 19 Mar 2014, at 16:46, jaflong jaflong <jafl...@yandex.com> wrote: > Hi, > > What are the requirements for connecting with tls/wss. > > I
Hi,
What are the requirements for connecting with tls/wss.
I have not come across any information or example for this.
My config is working when the client uses ws. However if I change this to use
wss, (this is it only paramter I change) it does not work.
I understand Kamailio does not support
Hi List,
Any suggestions on why I am getting the following issues.
I can get a successful tls connection when I connect with http
Tested by having this in kamailio.cfg
event_route[xhttp:request] {
set_reply_close();
set_reply_no_connect();
xhttp_reply("200", "OK", "text/html","Rec
Hi List
Is ice settings initiaised on the browser side or Kamailio websocket side?
Where are the setting parameters created for a=candidate
In the debug I get such logs as
a=candidate:3164634682 1 tcp 1509957375 xxx.xxx.xxx.xxx 0 typ host generation 0
a=candidate:3164634682 2 tcp 1509957375 xx
I am having problems with calls from webrtc to kamailio forwarded to Asterisk
These are snippet of the debug logs
Asterisk
CSeq: 4910 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
Jssip
Cause: Bad Media Descr
I have this in my config
$ru = "sip:"+$rU+"10.31.2.101:5090;transport=tls";
if (!t_relay_to_tls()) {
sl_reply_error();
return;
}
My ip address dest is 10.31.2.101.
Why has 130.31.2.101 been used?
13(10882) DEBUG: [ip_addr.c:243]: print_ip(): tcpconn_new: new
This is my tls.cfg for server
[server:default]
method = TLSv1
verify_certificate = no
require_certificate = no
private_key = /etc/asterisk/certs/proxy.key
certificate = /etc/asterisk/certs/proxy.crt
As far as I understand (verify_certificate = no), and (require_certificate =
no) should allow
Hi Everyone,
I want kamailio to do a single task, that is relay incoming (webrtc sip over
websockets) connections to asterisk. All sip handling is to be done is asterisk.
I have configured asterisk with tls settings.
My sample conf file is below
At the moment I am getting these errors
ERROR: t
Can anyone suggest why t_relay_to_tls cannot be found
0(15284) ERROR: [cfg.y:3272]: yyparse(): cfg. parser: failed to find
command t_relay_to_tls
i am using kamailio 4.1.1
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