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
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]
On 18/12/14 22:09, James Cloos wrote:
DM == Daniel-Constantin Mierla mico...@gmail.com 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
That's how I ended up going. It's working now. Thanks.
On Thu, Dec 18, 2014 at 4:11 PM, James Cloos cl...@jhcloos.com wrote:
MS == Marc Soda ms...@coredial.com writes:
MS I'm having a problem reassembling UDP packets on my Asterisk servers
after
MS passing through Kamailio
You could
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
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
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 rfc3264 is
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 be left in
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 the
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 SDP
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
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
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
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
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 to. After
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 rtpengine
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 rfu...@sipwise.com wrote:
On 12/19/14 10:47, Marc
DM == Daniel-Constantin Mierla mico...@gmail.com 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
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 table !
So
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
On 19 Dec 2014, at 23:35, Mahmoud Ramadan Ali cisco.and.more.b...@gmail.com
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
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