I tested this example and it worked for me. I was misunderstanding some of
the values and thought they applied to users being registered or not.
I am still uncertain what setting applies to allowing all traffic inbound
from a particular address. Any tips on this would be appreciated.
Thanks,
Ed
Daniel-Constantin Mierla writes:
> But in this case the call is completed, because negative response codes
> are absorbed by tm, 200ok being sent back, so no need to destroy any rtp
> session. iirc, for all branches of a parallel fork there is one rtp
> session in rtpproxy (cannot tell about ot
On 10/19/12 9:11 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
why is that? in failure_route I call rtpproxy_mange() which calls
unforce_rtp_proxy() which destroys the initiated session.
as explained in earlier messages of this thread, the situation unused
rtp sessions pile up happ
Daniel-Constantin Mierla writes:
> why is that? in failure_route I call rtpproxy_mange() which calls
> unforce_rtp_proxy() which destroys the initiated session.
as explained in earlier messages of this thread, the situation unused
rtp sessions pile up happens, e.g., when call parallel forks and
On 10/19/12 8:30 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
Popping in late, maybe I missed some parts, but I want to clarify that
all these cases discussed here were not tested with rtpproxy
application, right?
yes, they are all tested with real mediaproxy-ng rtpproxy server.
Daniel-Constantin Mierla writes:
> Popping in late, maybe I missed some parts, but I want to clarify that
> all these cases discussed here were not tested with rtpproxy
> application, right?
yes, they are all tested with real mediaproxy-ng rtpproxy server.
> For the archive, default config fil
Popping in late, maybe I missed some parts, but I want to clarify that
all these cases discussed here were not tested with rtpproxy
application, right?
For the archive, default config file destroys the rtp relaying session
in failure_route, when all branches are completed, doing it in onreply
I was looking at this logic some more and wanted to clarify my
configuration. I need to base the R-URI rewrite and MESSAGE forward based
on whether the account exists as a local subscriber or not rather than
whether the user is registered or not. This is connecting to an SMS
provider using SIP.
I
Hello guys.
In the Kamailio v.3.3.0 release notes
http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ it says:
all IPv6 network interfaces are auto-detected and Kamailio start listening
on them (if no strict listening rules are set)
But it doesn't happen.
version: kamailio 3.3.1 (x86_64/li
You can strip 100rel as long as it's Supported. However, not so good things
happen if you strip it from Required. :-)
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might
expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponc
Richard Fuchs writes:
> There's two timeouts, configurable through command line options, one is
> for active calls and defaults to 60 seconds, the other one is for
> silenced and not fully established calls and defaults to 1 hour. Calls
> will be cleared if no RTP traffic nor any re-invite has bee
Can I safely strip 100rel support from INVITEs sent by callers registered to my
proxy?
We're using proxy-generated 302 redirects for call forwarding in our
deployment. In general this works better for us than appending branches, as we
rapidly ran into branch limits when dealing with large multi
On 10/19/12 11:16, Juha Heinanen wrote:
> i have not see in any document description about how long rtp proxy
> keeps the call state after it has received US command, but no matching
> LS command. is there a timer that clears those hanging calls once in a
> while and sip proxy config writer does
Richard Fuchs writes:
> I suppose it could make sense to change mediaproxy's behaviour to ignore
> the to-tag given in the delete message if it hadn't received a lookup
> for that particular branch yet.
richard,
thanks for your reply. that would solve the problem that currently there
is no way t
Hi,
While I can't really answer your question, the logic in mediaproxy-ng is
that if the to-tag is given in the "D"elete message, it has to match the
to-tag that was previously given in the "L"ookup message alongside with
the from-tag. If no to-tag is given in the delete message, then only the
fro
Will this also permit incoming messages from this domain? Do I need to add
the gateway in any other settings besides the one you outlined?
Thanks,
Ed
On Fri, Oct 19, 2012 at 6:17 AM, Vitaliy Aleksandrov wrote:
> If I understood you right you just need to rewrite R-URI domain and
> forward M
i made more tests on deleting rtpprxy session when 480 is received. in
this test there is only one uas registered for AoR t...@test.fi. when
it times out and replies with 480, sip proxy makes exactly same call
rtpproxy_manage("FROW3");
first in onreply route and then i failure route. t
Hi,
MSRP AUTH requests must be authenticated using HTTP Digest
authentication. Part of the concatenated data when doing this
authentication is the method name. Because of the way MSRP requests are
formatted, the part of the request-line that would contain the method
name in HTTP or SIP requests
Juha Heinanen writes:
> Oct 19 15:50:15 siika mediaproxy-ng[12832]: Got valid command from
> udp:127.0.0.1:56183: 18594_12 D
> ptonjivixvat...@siika.tutpro.com;z9hG4bKqxklorkm wnzdf shsqn
> Oct 19 15:50:15 siika mediaproxy-ng[12832]:
> [ptonjivixvat...@siika.tutpro.com] Tags didn't match for de
i made rtpproxy test in setup where two sip phones have registered the
same AoR t...@test.fi. one is behind nat and the other is not. when i
call this AoR, my sip proxy executes
rtpproxy_manage("FROW3");
in branch route of the branch that is behind nat.
syslog shows:
Oct 19 15:49:59 siika /us
Dears,
I've changed the script to the following:
if (is_method("INVITE"))
{
if (!load_gws(1, $rU, $fu)) {
sl_send_reply("502", "Unable To lOad GatEwAyS");
exit; }
else while(next_gw())
If I understood you right you just need to rewrite R-URI domain and
forward MESSAGE if a user in not registered.
if (!lookup("location")) {
switch($retcode) {
case -1:
$rd = "gatewaydomain.com";
t_relay();
exit;
default:
sl_send
On 10/19/12 12:02 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
I fixed it on the master branch for now, probably it has to be
backported.
ok, thanks. looks like i'm the only one who has ever tried and for me
fix in master branch is enough.
I guess Andreas was the one that added
Daniel-Constantin Mierla writes:
> I fixed it on the master branch for now, probably it has to be
> backported.
ok, thanks. looks like i'm the only one who has ever tried and for me
fix in master branch is enough.
-- juha
___
SIP Express Router (SER)
Hello,
you would get better answer if you paste here the line 817 instead of
the whole config - it is not convenient always to save the file and open
it locally.
Cheers,
DAniel
On 10/19/12 2:16 AM, Ed Brady wrote:
Hi,
I am having trouble configuring MSILO. I copied the example config
into
Hello,
I fixed it on the master branch for now, probably it has to be backported.
Cheers,
Daniel
On 10/19/12 11:04 AM, Juha Heinanen wrote:
according to rtpproxy module readme, rtpproxy_destroy function takes
optional flags param:
5.4. rtpproxy_destroy([flags])
however, the code does not su
according to rtpproxy module readme, rtpproxy_destroy function takes
optional flags param:
5.4. rtpproxy_destroy([flags])
however, the code does not support it:
{"unforce_rtp_proxy", (cmd_function)unforce_rtp_proxy_f,0,
0, 0,
ANY_ROUTE},
{"rt
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