On Thu, Feb 6, 2014 at 3:26 AM, jaflong jaflong wrote:
>
>
> 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 under
On 02/06/14 14:42, Muhammad Shahzad wrote:
> Great, i would test Bundle right away. Just wondering if this branch
> also supports DTLS--SRTP. I would love to test that feature when available.
Not quite yet, but it's being implemented as we speak.
cheers
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Hi,
I need to get call-id and from-tag from a call in ringing state (2).
If I use dlg_manage() and put the call into a profile using
set_dlg_profile("usr","1").
Then do a avp_db_query() I don't see the call until it is connected.
I am using db_mode 1.
Yet if I do a sercmd dlg.dlg_list I
Great, i would test Bundle right away. Just wondering if this branch also
supports DTLS--SRTP. I would love to test that feature when available.
Thank you.
On Thu, Feb 6, 2014 at 2:43 PM, Richard Fuchs wrote:
> Hey,
>
> What mediaproxy-ng can do (which other RTP proxies don't do) is
> transl
Thanks por the reply,
it's like you suspect, in this scenario the kamailio proxy do some SBC
functions.
I'm doing tests with mhomed parameter. I'll do a feedback after these tests.
regards.
Diego
2014-02-05 14:47 GMT-03:00 Klaus Darilion :
> I am a bit confused. I suspect with SBC you mean
Daniel, Klaus,
Sorry to reply late to your comments, was busy doing some other things
until now.
Just provided some comments in
https://sip-router.org/tracker/index.php?do=details&task_id=380
Please take a look and let me know if you agree with my assessment. Thanks,
Ding
___
Integration - works.
Problem - dialing peer to peer via Kamailio OK but with missing VARs and
extension number, on dialing/transferring.
Maybe you know other way to configure Asterisk dialplan for users,
registered on kamailio and alowing dial as SIP/UserNumber insted SIP/
usernum...@kamailio.host.
To conclude this thread, the discussion was restarted on sr-dev mailing
list and a reworked patch is now in master branch.
Cheers,
Daniel
On 09/12/13 13:23, jay binks wrote:
Ok so... yea I should have waited another 10 min before posting to the
list..
I did the patch... it works for me.
if
Hi all!
Please give expanded tips about usrloc preload(). What exactly it
doing?
modparam("usrloc", "db_mode", 0)
modparam("usrloc", "preload", "location")
This case kamailio load from db on startup and operate in memory after?
--
WBR, Victor
JID: coy...@bks.tv
JID: coy...@bryansktel.ru
Hey,
What mediaproxy-ng can do (which other RTP proxies don't do) is
translate RTP/AVP (from regular SIP endpoints) to and from RTP/SAVPF
(encrypted RTP from WebRTC), which is what people usually use it for.
Assuming that ICE does its job correctly, two WebRTC endpoints should be
able to communic
Hi,
so with this heavy use of the usleep(); command i have to set the
"children=" parameter to a very high value.
Calculating with expected CAPS of 30 and up to 7 subsequent INVITES per
call and the sleep of 2 seconds, it's:
20*7*2 = 280
So it has to be
children=280
Do you see a Problem with thi
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
True, and actually not having it realtime is not a drama.
If you're having simultaneous calls in the hundreds, you don't care what's
happening *right* now, as long as you know what was happening in the recent
past, like 1-2 minutes delay, which is still good enough to monitor for
security issues p
On Thursday 06 February 2014 08:47:57 davy wrote:
> Would this be a good way to do this, or am I bluntly missing the better
> way?
Maybe just doing it "offline"? I just take the standard mysql acc accounting,
every minute I'll scan acc for completed calls, generate a csv, delete the
processed ac
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