Hello,
There is a message Possible Security issue with Kamailio - Asterisk
Realtime integration in Asterisk users mailing list:
http://lists.digium.com/pipermail/asterisk-users/2013-February/277633.html
I think the problem I have is somewhat similar.
Should I suppose that there is a
Hello,
Try allow allowguest=no in sip.conf [general] context and create a peer for
kamailio in sip.comf
Regards
Cibin
On 17-Jul-2014, at 12:52 pm, g.aloi...@gmail.com wrote:
Hello,
There is a message Possible Security issue with Kamailio - Asterisk Realtime
integration in Asterisk
Hello,
Try allow allowguest=no in sip.conf [general] context and create a peer for
kamailio in sip.comf
Regards
Cibin
On 17-Jul-2014, at 12:52 pm, g.aloi...@gmail.com wrote:
Hello,
There is a message Possible Security issue with Kamailio - Asterisk Realtime
integration in Asterisk
Hello,
I have:
allowguest=no
contactpermit=kamailio.ip.addr.ess
I also have tried the approach that I have peer kamailio, but then all
calls seems to go to to the context defined for kamailio peer. I do not
know how I could in that case handle individual calls - for example
determine if
Hello,
the spec was maintained by Peter (added here as explicit recipient) --
he used to build rpms, not sure about the current state. Maybe he can
add some comments about.
Cheers,
Daniel
On 16/07/14 23:13, Allen Zhang wrote:
Hi,
We cloned the kamailio 4.0.0 repository and the spec file
Hello,
have you looked at sip trace and checked what are the IP addresses in
the SDP? Maybe you need to swap the flags i and e.
You can eventually provide here the incoming invite as well as outgoing
invite, saying what you would expect to be in the outgoing one, so we
can give further
On Jul 16, 2014, at 4:05 PM, Andras FOGARASI fogar...@fogarasi.com wrote:
On 7/16/14, 10:00 PM, Frank Carmickle wrote:
On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
I expect that the signaling is ok at least for call setup.
From signling
Hi,
We did have a working msilo implementation a while back, but we don’t use it
any more, so I can only look back at the config files (which are a lot more
complicated than yours).
We added the a modparam modparam(msilo, outbound_proxy,
sip:MY_REAL_IP:MY_SIP_PORT;transport=tcp)
which will
Thanks for helping out Hugh (and obviously Daniel)
I have made some progress by adding the modparam (msilo,
outbound_proxy, sip:my.domain.com:5061;transport=tls).
Now indeed the stored message gets delivered when a UAC registers.
But I just noticed that it was also filling up msilo db with user
Hello,
I don't understand the patch you sent me. Is it for the PAI problem?
Regards,
Igor
2014-07-07 12:40 GMT+02:00 Igor Potjevlesch igor.potjevle...@gmail.com:
Hello,
Can you explain the modification and the impact on our plateform?
Is it for the pai problem?
Do you have explanation
On 7/17/14, 3:41 PM, Frank Carmickle wrote:
On Jul 16, 2014, at 4:05 PM, Andras FOGARASI fogar...@fogarasi.com wrote:
On 7/16/14, 10:00 PM, Frank Carmickle wrote:
On Jul 16, 2014, at 3:54 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
I expect that the signaling is ok
On Jul 17, 2014, at 11:27 AM, Andras FOGARASI fogar...@fogarasi.com wrote:
On 7/17/14, 3:41 PM, Frank Carmickle wrote:
I would expect that if it was a NAT issue you would see it much sooner than
15 minutes, 30-60 seconds. Are session timers being stripped by Kamailio?
You say it's a
On 7/17/14, 5:34 PM, Frank Carmickle wrote:
On Jul 17, 2014, at 11:27 AM, Andras FOGARASI fogar...@fogarasi.com wrote:
On 7/17/14, 3:41 PM, Frank Carmickle wrote:
I would expect that if it was a NAT issue you would see it much sooner than
15 minutes, 30-60 seconds. Are session timers
I have created an environment with the same config and I find the same problem.
While still does not work for video, I have changed (flip) the public/internal
IP addresses on rtpproxy and I can get half call leg working properly,
includding video.
However, I am testing video calls. So I got
Hello,
On 17/07/14 18:41, Igor Potjevlesch wrote:
Hello,
When this patch will be add in a new release? I can't try without
validation of new release.
when is scheduled the next release?
you were the only one reporting this issue. The patch will be backported
if you can test and confirm
On 17/07/14 23:10, Moacir Ferreira wrote:
I have created an environment with the same config and I find the same
problem. While still does not work for video, I have changed (flip)
the public/internal IP addresses on rtpproxy and I can get half call
leg working properly, includding video.
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