response hash sent by client, no it is not possible
> to decrypt it (at least under normal circumstance). You may find ways to
> modify the response hash, but it would be most likely pointless (since you
> do not know what was actually entered by the user as password).
>
> Th
Hello all,
During authentication, is there any way to affect the password user is
sending? I do suspect not as it is a clear security matter, but won't hurt
to ask. I use auth_db module with calculate_ha1 parameter set to 1. For
reasons in integrating Kamailio into my system architecture there is
Hello,
Something I've been wondering about meddling with sip uris with Kamailio:
I know it's possible to translate between a number prefix and a domain
using PDT, but is this possible: Having a numeric or alphanumeric value
stored in db, associated to a domain and appended to / removed from a
num
Hello,
A question on Kamailio variables and using dispatcher:
When in failure_route I want to know if the request message was going to a
dispatcher ip or a sip client ip (as in any other than dispatcher ip), how
do I make an if statement for that?
If I use ds_is_from_list(), I get wrong results
Hi,
The source for this string to int conversion error was found, it was just a
minor glitch in an if statement! Man, I feel stupid...
Anyways, the problem about calls not going through still persists. This I
located to the rtpengine_offer() call in a branch route. The sdp is not
changed and this
18:40 GMT+03:00 Olli Heiskanen :
> Hello,
>
> As outcome to my earlier sdp/rtp challenges I've upgraded my Asterisk
> version to 11.11.0 and still use a realtime integration with Kamailio. Now
> I face a somewhat different problem. With my setup I also changed from
> jssip cli
Hello,
As outcome to my earlier sdp/rtp challenges I've upgraded my Asterisk
version to 11.11.0 and still use a realtime integration with Kamailio. Now
I face a somewhat different problem. With my setup I also changed from
jssip client to a sip.js client in my websocket implementation. I cloned
th
Hello,
I don't know if this helps but I noticed you have a log entry:
Unknown flag encountered: 'force'
This is because rtpengine does not support this flag any more, it's
mentioned in the rtpengine module documentation:
http://kamailio.org/docs/modules/devel/modules/rtpengine.html#rtpengine.f.rt
http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f
Thank you Richard and everyone for helping. Even though the original
problem was never solved, all this has been extremely useful and
interesting.
cheers,
Olli
2014-07-31 20:28 GMT+03:00 Olli Heiskanen :
> Hi
ng again I can focus on the sdp side.
cheers,
Olli
2014-07-24 16:44 GMT+03:00 Richard Fuchs :
> On 24/07/14 09:27 AM, Olli Heiskanen wrote:
>
>>
>> That's odd... I pulled a new version from git master 4 days ago, and
>> copied the compiled rtpengine to /usr/sbin,
etc...), or maybe the way I've written my config?
cheers,
Olli
2014-07-23 18:13 GMT+03:00 Richard Fuchs :
> On 07/23/14 11:01, Olli Heiskanen wrote:
> >
> > Thanks,
> >
> > I think here's all of the call from before the called party answers:
> ...
>
>
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
$du = $null;
route(TOVOICEMAIL);
exit;
}
#!endif
Hi,
Thanks very much for this, that solved the double-m-line issue. Now I'm
calling rtpengine_offer in a branch route.
One issue still remains; the call still gets connected to the called zoiper
client, but it gets hung up right away. I traced this to be caused by a BYE
message from Kamailio, whi
TE going to the called client has nice clean rtp with RTP/AVP
profile.
I installed the latest rtpengine today. One thing I noticed the command
rtpengine --version output is undefined. Just a minor thing but good to
know.
Thank you for all your wonderful effort so far! Please let me know if you
need
clients.
cheers,
Olli
2014-07-12 17:27 GMT+03:00 Olli Heiskanen :
>
> Hello,
>
> I've started playing with an idea to add multiple asterisk servers and
> using dispatcher to balance the sip load between them. I added the code
> according to dispatcher module
Hello,
I've started playing with an idea to add multiple asterisk servers and
using dispatcher to balance the sip load between them. I added the code
according to dispatcher module documentation (
http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I
think there's something of
Hello,
Thanks for your suggestion, unfortunately it had no effect on the outcome.
This (using asterisk-kamailio integration with a domain specified for
clients) must have been achieved before, I wonder if I'm doing something
wrong here, or is this just not doable?
Thanks,
Olli
2014-05-18 21:29
) seems to send the message to what is
defined in the request line of the message so replacing it with 'testers.com'
would not work.
cheers,
Olli
2014-04-23 17:31 GMT+03:00 Pedro Niño :
> Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your
> sip.conf (aster
user, host, sippasswd, fromuser,
> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
> testers.com');
>
> --
> El abr 19, 2014 1:17 PM, "Olli Heiskanen"
> escribió:
>
>>
Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Expires: 3600.
Contact: ;expires=3600.
Date: Sun, 20 Apr 2014 09:04:41 GMT.
Content-Length: 0.
.
cheers,
Olli
2014-04-20 11:30 GMT+03:00 Mikko Lehto :
> 2014
Hello,
One of the tests I've been working with is Asterisk realtime integration
according to Daniel's guide here:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Weird thing is the client looks registered but I'm not sure if it really is
registered. If I'm not mistaken
Hello,
I'm probably still doing something wrong, I still get 488 from the
grandstream. Also zoiper refuses the call with 415 Unsupported Media Type.
According to the module description I tried to change my config to this:
Btw, thanks for enabling verbose flags, those are more readable when
workin
Hello,
After some tests, I'm still having some strange results.
When calling from ws client to grandstream, I get the below output to
/var/log/messages.
In a sip trace after 488 there are only INVITEs from kamailio server to
grandstream but no responses come back to kamailio server.
I haven't ch
Hi,
Thanks, it compiled nicely, I'll continue with more testing tomorrow.
- Olli
2014-04-08 15:36 GMT+03:00 Richard Fuchs :
> On 04/08/14 03:00, Olli Heiskanen wrote:
> > Hello,
> >
> > Thanks Juha, that will be a good thing to investigate more when I get my
>
1961: warning: format '%lu' expects type 'long unsigned int', but
argument 9 has type 'u_int64_t'
call.c:1961: warning: format '%lu' expects type 'long unsigned int', but
argument 10 has type 'u_int64_t'
make[1]: *** [call.o] Error 1
make[1]:
Hello,
Thanks, I'll look into the rtpengine, had a busy weekend but next week I'll
have better time.
The function seems like a good idea. I'd definetely rather use that if/when
it's available.
cheers,
Olli
2014-04-04 19:12 GMT+03:00 Juha Heinanen :
> Olli Hei
Hello,
Thanks, I'll give that a try and post back. I guess I install and run it
just like mediaproxy-ng?
I'll also try different sip clients like zoiper etc.
One thing that occurred to me based on the fact that the sdp is faulty, as
I did this test from the slides here:
http://www.slideshare.net
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
28 matches
Mail list logo
