On 19 Dec 2014, at 23:35, Mahmoud Ramadan Ali
wrote:
> Hi Dears,
> I'm a little bit confused about the difference between the "Subscriber" table
> and the "Location" table.I read that the "Location" table is used for
> persistent user registration BUT i did NOT configured the "Location" table
The users in subscriber table are the actual users who are allowed to
register to your SIP service. This is where kamailio gets the
authentication information, e.g. username and password etc.
The location table is where kamailio stores currently registered i.e.
online users. Obviously the records
Hi Dears,
I'm a little bit confused about the difference between the "Subscriber"
table and the "Location" table.I read that the "Location" table is used for
persistent user registration BUT i did NOT configured the "Location" table
and can get persistent user registration with the "Subscriber" tab
> "DM" == Daniel-Constantin Mierla writes:
DM> Can you elaborate? Otherwise I don't see what's the role of this reply,
DM> because that was clear they want want tls for postgres.
Apologies that I wasn't clear.
DM> The error message says 'no pg_hba.conf entry for host "..."' -- sounds
DM> li
Thanks for the response. You're right, the media stream is making it all
the way back to my PC, I just don't hear anything. And yes, my speakers
are turned up...
I'm not sure what to try next...
On Fri, Dec 19, 2014 at 12:31 PM, Richard Fuchs wrote:
>
> On 12/19/14 10:47, Marc Soda wrote:
> >
On 12/19/14 10:47, Marc Soda wrote:
> I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not
> getting audio back to my browser. From a packet capture I can see media
> from the browser to rtpengine, and then bi-directional RTP back and
> forth from my asterisk server, but rtpengin
On 12/19/14 11:39, Juha Heinanen wrote:
> Richard Fuchs writes:
>
>> I don't see how it would make a difference. If Firefox sends 0.0.0.0 and
>> rtpengine replaces it with its own address, then the receiving client
>> can send media to rtpengine, but rtpengine would have nowhere to forward
>> it t
Hello,
I'm using Kamailio with a SIP Application Server, when a user
registers on the IMS Core a 3rd Party REGISTER request is sent to the
application server to start some logic.
I'm trying to add in the 3rd party register request body the initial
REGISTER request sent by user device, and the 200
Even stranger, I get a media stream back to the browser when I use Chrome
(the first was with Firefox), but I still hear nothing. Also I get errors
like this in the log:
SRTP output wanted, but no crypto suite was negotiated
Full output:
https://gist.github.com/marcantonio/6c5414aa931a8f1c0072
Richard Fuchs writes:
> I don't see how it would make a difference. If Firefox sends 0.0.0.0 and
> rtpengine replaces it with its own address, then the receiving client
> can send media to rtpengine, but rtpengine would have nowhere to forward
> it to. After the answer, ICE processing may commence
I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not
getting audio back to my browser. From a packet capture I can see media
from the browser to rtpengine, and then bi-directional RTP back and forth
from my asterisk server, but rtpengine is not sending the media on to the
browser
On 12/19/14 10:02, Juha Heinanen wrote:
> Richard Fuchs writes:
>
>> Yes I understand, but 1) the mechanism of using 0.0.0.0 to put a call on
>> hold must remain operational and intact for those clients which use it,
>> and 2) if the offering client sends 0.0.0.0 in the SDP, then the
>> rewritten
Richard Fuchs writes:
> Yes I understand, but 1) the mechanism of using 0.0.0.0 to put a call on
> hold must remain operational and intact for those clients which use it,
> and 2) if the offering client sends 0.0.0.0 in the SDP, then the
> rewritten SDP should also contain 0.0.0.0, no matter what
On 12/19/14 09:33, Juha Heinanen wrote:
> Richard Fuchs writes:
>
>> On 12/19/14 03:32, Juha Heinanen wrote:
>>> i got mozilla to generate sdp with sendrecv, but still rtpengine does
>>> not replace 0.0.0.0 address on o and c lines. why?
>>
>> Because 0.0.0.0 means steam is on hold and so should
Richard Fuchs writes:
> On 12/19/14 03:32, Juha Heinanen wrote:
> > i got mozilla to generate sdp with sendrecv, but still rtpengine does
> > not replace 0.0.0.0 address on o and c lines. why?
>
> Because 0.0.0.0 means steam is on hold and so should be left in place.
what i understand from rfc3
Hi, DanB!
Kamailio has radius accounting too.
--
WBR, Victor
I use FREE operation system: 3.15.9- GNU/Linux
up 2 weeks, 1 day, 5 hours, 22 minutes
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-rout
On 12/19/14 03:32, Juha Heinanen wrote:
> i got mozilla to generate sdp with sendrecv, but still rtpengine does
> not replace 0.0.0.0 address on o and c lines. why?
Because 0.0.0.0 means steam is on hold and so should be left in place.
cheers
___
SIP
That's how I ended up going. It's working now. Thanks.
On Thu, Dec 18, 2014 at 4:11 PM, James Cloos wrote:
>
> > "MS" == Marc Soda writes:
>
> MS> I'm having a problem reassembling UDP packets on my Asterisk servers
> after
> MS> passing through Kamailio
>
> You could try having the kama->
On 18/12/14 22:09, James Cloos wrote:
>> "DM" == Daniel-Constantin Mierla writes:
> DM> The question would be more specific to the error message printed from
> DM> postgres client library:
>
> DM> FATAL: no pg_hba.conf entry for host "129.240.1.1", user
> DM> "foo_test_user", database " foo_
Hi All,
I am running ims servers(pcscf,scscf,icscf and hss) as part of kamailio
proxy.
And I am trying to register and unregister the end-points, with help osip
lib.
But I see the below ERROR message.
ERROR: *** cfgtrace:failure_route=[REGISTER_failure]
c=[/etc/kamailio/pcscf/kamailio.cfg] l=90
i got mozilla to generate sdp with sendrecv, but still rtpengine does
not replace 0.0.0.0 address on o and c lines. why?
-- juha
Dec 19 10:20:18 box /usr/bin/sip-proxy[5841]: INFO: =
rtpengine_offer(ICE=force replace-session-connection replace-origin
via-branch=1)
Dec 19 10:20:18 box rtpen
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