Hi All,
I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting
below error though websocket.so is present in /usr/local/lib64/modules/
load_module(): ERROR: load_module: could not open module
: libunistring.so.0: cannot
open shared object file: No such file or directory
may I
Kelvin Chua writes:
> i was able to apply the RET_ARRAY fix to htable.dump in 4.0.5
> however unable to share the patch as htable_rpc[] is nowhere to be found on
> trunk
> was there any changes from 4.0.x to trunk with regards to rpc?
see
rpc_export_t htable_rpc[] = {
{"htable.dump", hta
Looks like its crashing because a special character trying to be parsed,
maybe a character used for password?
Not sure, but try to use simple password (just for test) without special
characters and check if having crash again.
BTW, Daniel is busy, expect answer but be polite and have some patienc
I think you should remove this section: or comment it, its behavior is not
the one we want at this moment.
---
if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if
(is_method("OPTIONS")) { # send reply for each options request
sl_send_reply("200", "OK"); }
-
El abr 1, 2014
Sorry, I was out for a while. Still have this issue?
>From what I am seeing, asterisk is expecting for the password. Is the
voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the
voicemail. Have the extensions.conf at asterisk
Really have no idea if supports mongodb... Basic search got me no results.
Will take a deeper look, and provide a better answer.
El abr 1, 2014 2:42 AM, "Rizwan Khan" escribió:
> Thanks a lot Pedro.
>
> Just one question? Do we have module for Mongodb? how do i use it?
>
> Rizwan Khan
>
>
>
>
> O
i figured out why loose_route() did not work as expected. it is because
the party that sent the bye, had messed up its request uri and replaced the
host:port of remote target with host:port of the proxy causing
loose_route() to work in strict mode.
teaching: if you see these headers in a request
Hey,
Unfortunately your packet dumps are truncated and don't show the
complete SDP bodies. It would also be interesting to see which options
and parameters are passed to mediaproxy-ng when processing the SDP. You
would find this info in the log produced by mediaproxy-ng, which should
also include
my proxy is listening at udp/tcp ip:port 192.98.102.30:5060 and my
script has this for in-dialog requests:
route [IN_DIALOG_REQUEST] { # handle in-dialog request
if (!loose_route()) {
xlog("L_INFO", "Loose route failed on $hdr(route)\n");
...
exit;
};
xlog("L_
Hello,
I've been experimenting with Kamailio with ws and sip clients and could
need a hand in getting a call between those two to work.
I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS
6.5 and a mediaproxy-ng running. I have clients wscli...@testers.com and
gscli...@teste
Hello Sirs,
I managed to make it work but with the resource list embedded in the
rls-services xml.
The xml looks like that:
http://www.w3.org/2001/XMLSchema-instance";>
presence
Now, I'm trying to make it work using an
Hi all,
I am trying to do an on-registration to a Freeswitch box, to to this I am
trying to use
uac_reg_request_to(user, mode)
Does anyone have any examples with this working? The request is going to
the freeswitch box OK, I can see the challenge header being built but it is
not sent back to the f
You might need to also add asterisk 12 b2b in order to convert to simple
sip to solve issues with ice on the same box.
On Apr 1, 2014 11:52 AM, "ik" wrote:
> Hello,
>
> I'm a newbie with Kamailio, and I require to connect webrtc (websockets)
> based phones, into a VoIP PBX that does not support w
Hello Daniel,
I fixed config and right now it fully functional local calls. VM server still
need see why asterisk not send correct direction rtp.
Here config.
http://sebsauvage.net/paste/?ffca000a32579ed0#JhOA+vZnTWFH05V2Ey7lL6u5Ntp95jyg84EpBxe0sa0=
Slava.
- Original Message -
Hello,
I'm a newbie with Kamailio, and I require to connect webrtc (websockets)
based phones, into a VoIP PBX that does not support websockets.
I wish to create/use Kamailio rules that will translate UDP to websockets
and vice versa.
I have found few examples over the internet, but as it seems (
Hi , i have a question for all : in the onreply_route is visible a dialog
variable ($dlg_var(ring_time)) or a avp variable ?
Can I set them , on this kind of route ? On the event route (dialog start)
they are all null.
I try to set them, but some time (on reInvite) this variables are not
se
Hi,
I'm using kamailio as an intermediate proxy with survivability. That means I'm
monitoring the primary server using the dispatch module, and if it goes down, I
take control of all further requests. The main server acts as registrar, but I
keep a copy a of the registered users list.
There's
Thanks Carsten,
Are there so many things missing? Some instructions in the different routing
blocks (terminating, originating, ...)?
If you have a quick clue that could unblock me, you can tell me
Otherwise I'll wait till Eastern
Olivier
-Message d'origine-
De : sr-users-boun...@lis
Hello,
Do you receive my emails? And on the list?
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
Envoyé : lundi 31 mars 2014 18:34
À : mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] Crash on REGISTER
Hello,
Is the p
Hi Olivier,
i will update the configuration files for WebSockets at the P-CSCF
after KamailioWorld. The week after KamailioWorld, we are at another
congress, but probably around easter, you should find a proper
configuration for WebSocket on the P-CSCF (including media).
Kind regards,
Carsten
20
Hello all,
I have installed Kamailio 4.1.2 in IMS configuration (3 Kamailio instances to
run in P/I/S-CSCF modes)
Everything works fine with Native IMS clients (Boghe)
I want now to introduce Web clients using WebSockets. I have added to the
P-CSCF Kamailio.cfg the websockets parts
Registratio
Hello,
I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc
and SIP\rtp.
Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP
gateway (signaling, audio or video transcoding, ICE and so on)?
Some months ago I found the architecture described here
My next project is ...
Setting up a MySQL 5.6 plus 4x MySQL Cluster NDB 7.3 in memory data node
to increase the data throughput
Like this one:
http://dev.mysql.com/doc/refman/5.1/de/images/cluster-components-1.png
Regards
Rainer
Am 01.04.2014 09:12, schrieb Rizwan Khan:
Thanks a lot Pedro
Also, MongoDB is not a relational database like MySQL or PostgreSQL. The choice
to use a schemaless/NoSQL database vs. an RDBM must be rooted entirely in the
type of data you're storing and how you want to store it. These concepts are
not freely interchangeable.
On 1 April 2014 03:16:13 GMT-0
All modules are listed here:
http://kamailio.org/docs/modules/4.1.x/
On 1 April 2014 03:12:10 GMT-04:00, Rizwan Khan wrote:
>Thanks a lot Pedro.
>
>Just one question? Do we have module for Mongodb? how do i use it?
>
>Rizwan Khan
>
>
>
>
>On Mon, Mar 31, 2014 at 5:52 AM, Pedro Niño
>wrote:
>
Thanks a lot Pedro.
Just one question? Do we have module for Mongodb? how do i use it?
Rizwan Khan
On Mon, Mar 31, 2014 at 5:52 AM, Pedro Niño wrote:
> As Alex said, I/O and calls per second (CPS) is Dependant on what type of
> design you are using.
>
> As a tip, mysql is good for small to
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