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).
Thank you.
On Fri, Dec 26, 2014 at 7:33 PM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote
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
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
Olli Heiskanen ohjelmistoarkkite...@gmail.com:
Hi,
A little follow-up on this:
The problem only happens when I call rtpengine_offer() inside a
branch_route. If I call rtpengine_offer() in the failure_route (after 488)
this conversion error does not happen, but then I get the double sdp
GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
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
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:
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
://forums.digium.com/viewtopic.php?f=1t=90167sid=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 ohjelmistoarkkite
2014-07-24 16:44 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
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, which is running. (although
might help verifying the version if command
written my config?
cheers,
Olli
2014-07-23 18:13 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
On 07/23/14 11:01, Olli Heiskanen wrote:
Thanks,
I think here's all of the call from before the called party answers:
...
I can't seem to reproduce this, when I run through the same sequence
;src_ip=1.1.1.1;dst_ouser=771;dst_user=771;dst_domain=2.2.2.2
cheers,
Olli
2014-07-21 16:38 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
On 20/07/14 01:15 PM, Olli Heiskanen wrote:
Hi,
...
There may be something off in my Asterisk configs since it's Asterisk
that responds 488, but see
;
}
#!endif
#!ifdef WITH_WEBSOCKETS
route(UA_FAILURE);
#!endif
# ... dispatcher for 500 reply or local timeout handling
}
cheers,
Olli
2014-07-23 16:32 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
On 07/23/14 05:03, Olli Heiskanen wrote:
Hi,
Thanks very much
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 any logs or traces and I'll provide them asap.
cheers,
Olli
2014-04-21 17:18 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
On 04/12/14 09:31, Olli Heiskanen wrote
clients.
cheers,
Olli
2014-07-12 17:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
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
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
...@gmail.com:
Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your
sip.conf (asterisk) to show the realtime peers
El abr 23, 2014 8:29 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com
escribió:
Hello,
Gracias Pedro, kiitos Mikko.
It's good to know I have configured Kamailio
) VALUES ('660', '660', 'dynamic', 'password', '660', '
testers.com');
--
El abr 19, 2014 1:17 PM, Olli Heiskanen ohjelmistoarkkite...@gmail.com
escribió:
Hello,
One of the tests I've been working with is Asterisk realtime integration
according to Daniel's guide here:
http
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
] --
Media #1, port 30240 [::]:0, 0 p, 0 b, 0 e
Apr 12 17:15:37 u363id562 rtpengine[13839]: [dtgbbcrpth16rb2ks58c]
Returning to SIP proxy: d6:result2:oke
cheers,
Olli
2014-04-10 16:48 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
On 04/10/14 09:26, Olli Heiskanen wrote:
Hello
configuration properly.
Regards,
Peter
On 6 April 2014 19:58, Juha Heinanen j...@tutpro.com wrote:
Olli Heiskanen writes:
Thanks, I'll look into the rtpengine, had a busy weekend but next week
I'll
have better time.
what comes to peter's slideshare failure_route example, i think it only
', but
argument 10 has type 'u_int64_t'
make[1]: *** [call.o] Error 1
make[1]: Leaving directory `/usr/local/src/rtpengine/daemon'
make: *** [all] Error 2
cheers,
Olli
2014-04-06 21:58 GMT+03:00 Juha Heinanen j...@tutpro.com:
Olli Heiskanen writes:
Thanks, I'll look into the rtpengine, had a busy
Hi,
Thanks, it compiled nicely, I'll continue with more testing tomorrow.
- Olli
2014-04-08 15:36 GMT+03:00 Richard Fuchs rfu...@sipwise.com:
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
simple unrealistic
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 j...@tutpro.com:
Olli Heiskanen writes
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:
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
27 matches
Mail list logo
