:5060;branch=z9hG4bKbdb2.
> c896050feb9a2151548b9fe6c3f776fb.0.
> Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK-
> 524287-1---1034af445c791a63;rport=53487.
> Max-Forwards: 68.
> Route: .
> Contact: 37d7b85846910961>.
> To: "6093756295 <(609)%20375-6295>" ;
>
traversal,
except symmetric NAT why do we always use proxy option in case if NAT is
detected.
On Wed, Feb 8, 2017 at 12:01 PM, Daniel Tryba wrote:
> On Wed, Feb 08, 2017 at 01:12:05AM -0700, Arsen Semionov wrote:
> > good question from 2013 :)
> > Maybe someone has experience
hanks!
Arsen.
Khoa Pham wrote
> Hi all,
>
> When using STUN, I can detect my NAT type. The SDP contain x-NAT field (0:
> unknown, 1: full cone, ..., 6: symmetric) which tells Kamailio the NAT
> type
> of clients. Why doesn't Kamailio use that ?
>
> --
> K
easier by checking From/To header at the
asterisk side and route calls appropriately within a single dialplan
context.
Regards,
Arsen.
On Tue, Feb 7, 2017 at 9:36 AM, Tomas Zanet wrote:
> Hello, thanks to this guide
> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-
> asterisk-11.
don't even see an error
> from dialog module.
>
>
>
> Anyone has a better understanding of the way the module runs? Thank you!
>
>
>
> Regards,
>
>
>
> Igor.
>
> ___
> SIP Express Router (SER) and
Hi guys,
In addition to this interesting and useful thread, what is the best way to
implement media session recovery, for example in Active/Passive HA scenario?
I know that it is possible with rtpengine (redis db), is it possible with
rtpproxy?
Thanks,
Arsen.
On Wed, Oct 19, 2016 at 11:19 AM
from the debug messages?
>
> I guess you require client certificate in your config.
>
> Cheers,
> Daniel
>
> On 26/05/16 15:06, Arsen wrote:
>
> Hi guys!
>
> I am trying to configure kamailio with WSS.
> We have trusted certificate installed SIP over TCP/
se firefox, when I use chrome it
even doesn't show me an error..
Any ideas?
Thanks!
--
Regards,
Arsen.
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e kamailio there is no any data, so
our script is waiting until timeout...
Please advice.
Thanks,
--
Regards,
Arsen.
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Sorry forgot to specify (in-memory) not using the DB, I feel like using the DB
for such a task would be such a drag on it
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Since this thread is open, I wanted to ask, is it possible to replicate dialog
data over to multiple nodes as well?
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Richard that was a champion’s answer thank you so much! Maybe it should be
noted kamailio's rtpengine module documentation.
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Hi Richard! Here’s the output at log level 7
[1452404737.610730] WARNING: Failed to properly parse UDP command line '11683_0
d7:command4:pinge' from 10.0.0.10:54602, using fallback RE
[1452404737.619006] WARNING: Failed to properly parse UDP command line '11683_1
d7:command4:pinge' from 10.0.0.1
Hi everyone, i’ve been having this “not so problem” going on. So I have
rtpengine installed on a server and use the default rtpproxy module on kamailio
and it works beautifully. Having read the rtpengine modules description, I see
that it is a drop in replacement of rtpproxy. So I keep trying t
Alright I guess this is one of those duhh… moments, all this time I was
forgetting to add the content length to the NOTIFY event. Remark: DO NOT FORGET
the content length!
Thanks for the help, very much appreciated!
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I see, what i’m trying to do is actually to limit querying the DB by updating
the MWI only when there is activity.
In other words to send a NOTIFY when the user receives a new message in his
voicemailbox.
I am not sure if it’s possible to send the NOTIFY for example 5-30 mins after
the initial
Hi everyone, I’m trying to send a NOTIFY event using “sipsak” to enable the MWI
I read up a lot of documentation and didn’t really find the information needed
to accomplish this.
So I have:
Phone 1 -> P1
Server 1 -> S1 (Kamailio 4.3.4)
Server 2 -> S2 (sipsak)
P1 registers to S1 and creates a new
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