,"Busy Here");
exit;
}
Regards
Trevor Steyn
On 22/10/2015 15:59, Rodrigo Pimenta Carvalho wrote:
>
>
> Hi.
>
>
> In a test case with OpenSIPS, I have:
>
>
> C1 = caller 1.
>
> C2 = caller 2.
>
> D1 = callee 1.
>
> D2 = callee
Hi No you shouldn't have to ,but this depends on your rtpproxy version
whether it support autobridge as i understand you have a custom forked
rtpproxy.
To be safe i would just use the "ie" as you have the control in the
script to make the right decisions.
Regards
Trevor Steyn
O
HI Cedric,
As i understand using
modparam("rtpproxy", "rtpproxy_autobridge", 1)
Will let rtpproxy figure out which interface it needs to be offering the
session on based on the the IP address and port present in the command
frame.
Regards
Trevor Steyn
On 15/10/2015 17:
Hi,
Why are you using both mediaproxy and rtpproxy?
If you plan on using both which is possible you should at least not let
the rtp port ranges overlap.
can you show us where you are using rtp relays on your routing scripts.
Regards
Trevor Steyn
On 13/10/2015 08:48, chiu ching cheng wrote
Normally if you are in bridge mode you will use the "ie" flags which i
dont see please can you post how you start rtpproxy.
Regards
Trevor Steyn
On 01/10/2015 08:47, Raistlin Majere wrote:
> Hi again,
>
> Well I have read a bit more about the modes and yes, we use brigde
&g
are it with you just want to confirm your setup.
Regards
Trevor Steyn
also you dont use rtpproxy in re-invites this would cause re-invites to
skip rtpproxy im sure this is a mistake.
On 30/09/2015 17:05, Raistlin Majere wrote:
> First of all thanks for your quick response.
>
> At the mome
answer and you could remove the reply routes and just relay
if ($var(ciptrusted)=="yes") {
engage_rtp_proxy("focnr");
} else {
engage_rtpproxy("focn");
}
Regards
Trevor Steyn
On 30/09/2015 14:56, Raistlin Majere wrote:
> Hi,
>
> Recently the maintaine
Hi Peter,
I am using the 1.2.1 from epel, This version makes CPU usage go to 100%
and stay's there consistently which is worrying.
Is there a problem with the one from epel?
Regards
Trevor Steyn
On 14/08/2015 15:00, Peter Lemenkov wrote:
> Hello All!
>
> I'd say stay with 1
proxy/
But this version seems very old and Readme states its for Opnesips 1.6.4.
Any guidance on my issues would be very much appreciated.
Regards
Trevor Steyn
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Hi
The permissions module does this. The standard trunking script has support
for limiting channels.
Regards
Trevor Steyn
On 13 Aug 2015 6:14 PM, "Terrance Devor" wrote:
> Hello Everyone,
>
> Was wondering what is the preferred and current means of managing
> concurrent
Thanks Bogdan,
You have been a great help this is way better and cleaner way than what
i was going towards.
And thanks for the great project.
Regards
Trevor Steyn
On 05/08/2015 12:17, Bogdan-Andrei Iancu wrote:
> Hi Trevor,
>
> The proper (as syntax) way of doing the lookup
y using avp_subst like so
avp_subst("$avp(foo)/$avp(string_custom_route)/",
'/.*Route=[^"]*"([^"]*).*/\1/');
then i use ///$avp(string_custom_route)/ to route the call, I feel like
this is not good practice if you could guide me here i would appreciate it
as i hav
://www.opensips.org/html/docs/db/db-schema-devel.html#AEN3101
Drouting Module:
http://www.opensips.org/html/docs/modules/2.1.x/drouting.html
Regards
Trevor Steyn
On 23/07/2015 20:36, Kevin Kokos wrote:
> Hi All,
>
> First of all, i'm a noob in the sip environment, network and even more
eciated I would just like to be pointed in
the right (Best possible way to do this) direction.
Regards
Trevor Steyn
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raffic by adding another listen line in your global config
listen=udp:NEW_IP:5060
Regards
Trevor Steyn
On 27/07/2015 15:47, kevinfang wrote:
> HI,
> I have a opensips servers in a private network, IP address:
> 10.34.14.24, now I'm going to this private network address 1: 1 NA
Hi Ahmed,
Have you set
short_open_tag = On ;
In your php.ini file?
Also tailing your http error and access logs might give you some
insight to your problem.
Regards
Trevor Steyn
On 27/07/2015 16:15, alneami_ah...@yahoo.co.uk wrote:
> Hi all,
> Well I am a new to opensips and
Hi Anyone have any ideas on the below issue?
On 23/07/2015 17:12, Trevor Steyn wrote:
> Hi Guys,
>
> I seem to be having some trouble with the new topology_hiding module in
> opensips 2.1
>
> here is the call scenario
>
> UAC --> Opensips --> UAS
>
> UAC Send
060;user=phone]
Jul 23 16:01:16 [22780] DBG:core:get_hdr_field: to body
[]
Jul 23 16:01:16 [22780] DBG:core:get_hdr_field: cseq : <2>
Jul 23 16:01:16 [22780] DBG:maxfwd:is_maxfwd_present: value = 65
[Script Trace][/etc/opensips/opensips.cfg:185][me][core if] -> (BYE from
10.10.16.2, ruri=
; [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Trevor Steyn
> *Sent:* Thursday, July 23, 2015 11:27 AM
> *To:* users@lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] changing $rU number
>
>
>
> Hi Denis
>
> You can use the dialplan module to rewrite var
TCP stats: 2 in from callee, 5 in from caller, 7
relayed, 0 dropped
INFO:remove_session: session on ports 57596/33858 is cleaned up
DBUG:doreply: sending reply "16658_5 0
Regards
Trevor Steyn
On 23/07/2015 10:28, Rik Broers wrote:
> Try it with only the ie flags, wz20 only adds more c
Hi Denis
You can use the dialplan module to rewrite variables as below
dp_translate("1","$rU/$rU");
Then insert you regexp into the dialplan tables, read the dialplan
module documentation for more info.
Regards
Trevor
On 23/07/2015 09:49, dpa wrote:
>
> Thank you.
>
> But what can I use for do
"
>
> You could try and use the rtpproxy_offer and answer functions. put in the
> reply route an if (has_body("application/sdp")) to also catch the 183 with
> sdp .The docs have examples on how to use them and how to trigger on reply
> routes.
>
> Regards,
&
exit;
}
dp_translate("1","$rU/$rU");
# route calls based on prefix
if ( !do_routing("1""$var(gw_attributes)") ) {
send_reply("404","No Route found");
exit;
}
if (is_method("INVITE")) {
force_send_socket(udp:http://salamander.iburst.co.za:8000/personal/signalling.txt
I have tried a most of the options on rtpproxy_engage with no luck
Regards
Trevor Steyn
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