On Fri, Aug 29, 2014 at 5:08 PM, Alex Villacís Lasso
wrote:
> El 29/08/14 14:44, Paul Belanger escribió:
>
>> On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso
>> wrote:
>>>
>>> El 28/08/14 19:09, Paul Belanger escribió:
>>>
>>>&
On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso
wrote:
> El 28/08/14 19:09, Paul Belanger escribió:
>
>> On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso
>> wrote:
>>>
>>> As a continuation of my project, I am trying to set up Kamailio as a
&
s is that since the REGISTER has a contact with
> transport=ws , Asterisk wants to send this through a websocket (which is
> disabled). So I could have to generate a contact without transport=ws .
>
> I have worked around this by setting qualify=no in the account for the
> web
On Tue, Aug 26, 2014 at 9:21 PM, Richard Fuchs wrote:
> On 08/26/14 20:58, Alex Balashov wrote:
>> On 08/26/2014 08:56 PM, Paul Belanger wrote:
>>
>>> I'd agree 'drop-in' replacement is not correct. I ran into the same
>>> issues as you. Curre
The question is:
> can it be used to bridge DTLS-SRTP, without touching the (encrypted)
> payloads, and delegate the decryption to asterisk itself?
>
While my setup is not the same as yours, the fix would be to bind
asterisk to the same interface as kamailio (different port for SIP),
then
gt; config file only.
>
I wonder what would be needed to implement some sort of TTL for the
dispatcher cache. Do any of the other modules implement TTL in any
fashion for DB queries?
--
Paul Belanger | PolyBeacon, Inc.
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Git
On Tue, Aug 5, 2014 at 11:06 AM, Richard Fuchs wrote:
> On 05/08/14 11:01 AM, Paul Belanger wrote:
>>
>> I was hoping somebody could confirm the following is a _normal_ log
>> file for rtpengine. Specifically I am curious of the 'Successful STUN
>> binding request
On Tue, Aug 5, 2014 at 11:06 AM, Richard Fuchs wrote:
> On 05/08/14 11:01 AM, Paul Belanger wrote:
>>
>> I was hoping somebody could confirm the following is a _normal_ log
>> file for rtpengine. Specifically I am curious of the 'Successful STUN
>> binding request
sk because I am getting one way audio at the
moment. I get audio from rtpengine however rtpengine does not hear
me.
http://pastebin.com/hztxMFL3
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twit
On Mon, Aug 4, 2014 at 1:37 PM, Richard Fuchs wrote:
> On 04/08/14 01:10 PM, Paul Belanger wrote:
>>
>> Greetings,
>>
>> I'm having some trouble getting dtls-srtp working with kamailio 4.1
>> (mediaproxy-ng) and rtpengine (master).
>>
>> I believ
On Mon, Aug 4, 2014 at 1:37 PM, Richard Fuchs wrote:
> On 04/08/14 01:10 PM, Paul Belanger wrote:
>>
>> Greetings,
>>
>> I'm having some trouble getting dtls-srtp working with kamailio 4.1
>> (mediaproxy-ng) and rtpengine (master).
>>
>> I believ
ed is my debug log file.
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
Aug 4 17:05:22 webrtc-01-prod /usr/sbin/kamailio[8737]: WARNING: dispatcher
[dispatch.c
rtpengine and bind them to each interface? Additionally, in this
setup can audio be bridged between the 2 rtpengine processes?
If not, how do people over come this difference?
--
Paul Belanger | PolyBeacon, Inc.
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Github: https
Anybody know why deb.kamailio.org is down? Is there a mirror setup
some place else?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
On 13-10-22 01:36 PM, Paul Belanger wrote:
On 13-10-22 03:14 AM, Klaus Darilion wrote:
OF course you could just SUBSCRIBE to get NOTIFYs. But then you would
need to subscribe to all users (e.g. subscribe a user whenever there is
a new registration).
I think a cool feature would be a 'wil
SCRIBE sip:*@mydomain.com to receive all events of mydomain.com, or
SUBSCRIBE *@*.
Right, I was trying to avoid doing something like that, only because of
the overhead it creates. However, if that is the best solution I would
consider it.
The wildcard subscribe is a cool concept, however.
--
Pau
gure I see if
there is another method. EG: Opening a websocket to stream events?
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter: https://twitter.com/
On 13-03-07 05:30 PM, Alex Balashov wrote:
What gives you that idea? Most likely, they spoofed an IP.
Here is a example INVITE message.
http://pastebin.com/a1SSeyMk
--
Paul Belanger | PolyBeacon, Inc.
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Github: https
On Thu, Mar 7, 2013 at 5:24 PM, Alex Balashov wrote:
> Because digest authentication is a far from self-evident or universal
> use-case for Kamailio.
>
>
> Paul Belanger wrote:
>>
>> Greeting,
>>
>> Hopefully, I'm understanding the following default k
ain;f=etc/kamailio.cfg;hb=HEAD
[2] http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
_
me this? Perhaps
tagging their call control in some other method?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
___
SI
e Asterisk register to kamailio, I can route
calls fine however, I believe the dispatcher is the better way to do
this.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/
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