Hello:
A->opensips (+mediaproxy)-> B
Can any one help me why MediaProxy don't add Relay
candidate? I am sure that use_media_proxy works before opensips send
invite to B.
A(pjsip) Send Invite, B side receive Invite
include relay candidat
Hello Everyone,
We have been successful so far with our virutal machine prototpe:
2 VM for HA Mysql
2 VM for Asterisk
2 VM for OpenSIPS (in/out)
Now we would like to iinclude MediaProxy with the OpenSIPS-in VM
however, the webiste mentioned not to install MediaProxy on a VM?
This is just a proto
MediaProxy it is! Thank you sir.
Cheers,
Nick
On Wed, Nov 30, 2011 at 7:15 PM, wrote:
> I use Mediaproxy and don't have any issues with it. I have no experience
> with RTPProxy. OpenSIPS b2bua is not what you are looking for when it comes
> to RTP. The B2B_LOGIC module can do network hiding (b
I use Mediaproxy and don't have any issues with it. I have no experience
with RTPProxy. OpenSIPS b2bua is not what you are looking for when it comes
to RTP. The B2B_LOGIC module can do network hiding (b2b_init_request("top
hiding")) but so can the dialog module now (topology_hiding()).
With
Hey Duane,
Thank you so much for your response. That is exactly my problem.
Currently I only have
OpenSIPS flowing SIP packets.
As for the actual RTP, I was thinking of using either the STUN or
NATHELPER module.
Only I am not worried about NAT, I only need the flowing of RTP along
with SIP. What
Hi to all,
in our system we have more or less 1900 AOR registered.
We use the mi_xmlrpc module to query the proxy on the realtime AOR in the
system.
The issue is that for some records the mi_xmlrpc answer with AOR and Contact in
the same field and for other it splits in 2 different fields.
On 190
Are your diagrams the path that SIP takes or RTP? Asterisk can be internal
and private from the outside world when it comes to SIP. It can also be
internal only when it comes to RTP but you would need to use a relay server
like Mediaproxy or RTPProxy. Mediaproxy can sit on the internet with a
On Wed, Nov 30, 2011 at 3:05 PM, Jerry Richards
wrote:
> Hello,
>
> ** **
>
> I'm new to OpenSIPS and just trying to install it for the first time on my
> CentOS 5.3 machine, but I cannot register a phone. I did not change any of
> the default configurations.
>
> **...
>
> Can anyone t
Hello,
I'm new to OpenSIPS and just trying to install it for the first time on my
CentOS 5.3 machine, but I cannot register a phone. I did not change any of the
default configurations.
Here are the steps I followed based on the install doc:
1) Downloaded the source code for version opensips_1_
Hello Everyone,
We are trying to close the doors entirely to our asterisk servers, making
only opensips visible to the outside world:
Incoming -> OpenSIPS -> Asterisk -> OpenSIPS -> Trunk
| In Out |
| _ _ _ _ _ _
I see. does that work for 1.7 too?
On 30 November 2011 14:27, Razvan Crainea wrote:
> Hi, Pete!
>
> I think that registering an event notification using the event_datagram
> module requires a timeout, but the event_rabbitmq doesn't. But what you can
> do is to subscribe the command directly
Yes, it is possible.
Leave the existing opensips/mediaproxy as is and add a new
opensips/rtpproxy server.
The new server must listen on two interfaces (IPV4/IPV6) and rtpproxy
must be configured to run in bridge mode (see option -l IPV4/IPV6).
The rtpproxy commands must be invoked with proper "ei"/
Hi, Pete!
I think that registering an event notification using the event_datagram
module requires a timeout, but the event_rabbitmq doesn't. But what you
can do is to subscribe the command directly from OpenSIPS script. There
if you don't specify a timeout, the subscriber will be registered
f
One other quick question, the fifo for setting up the listeners specifies a
timeout...
is it possible to set that to infinity (or until opensips is stopped?)
On 30 November 2011 14:16, Pete Kelly wrote:
> Thanks Razvan.
>
> I used a really simply python script for this... don't know if you woul
Thanks Razvan.
I used a really simply python script for this... don't know if you would
want to include in the docs?
import socket
server_socket = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
server_socket.bind(("", 8085))
print "Server up on 8085"
while 1:
data, address = server_so
Hello,
We use opensips with Mediaproxy and CDRTool on our VoIP gateway.We will
be introducing IPv6 in our infrastructure too and I would also like to
have SIP connectivity available for IPv6. Our upstream SIP links are
IPv4 only, so we will need some kind of IPv6 to IPv4 gateway.
AFAIK opensips c
Hi, Pete!
In order to use OpenSIPS Event Interface, you need to create an external
application that listens for an event. This application should create
the UNIX socket used for communication, and then subscribe to OpenSIPS
for a certain event, using the event_subscribe MI command. When the
e
Hi
I am using opensips 1.7 and would like to raise an event when I see a BYE,
however I am struggling with the documentation as it stands. This is what I
am doing:
In the script:
if(is_method("BYE")) {
xlog("BYE seen, raising event");
$avp(s:attr) = "number";
$avp(s:val) = 0;
$avp(s:attr) = "s
Hi
Yes this module would be really useful for us if the VHost / exchange /
routing key could be specified independently for the messages sent.
Maybe the routing key could be the message name in the raise event? e.g.
E_PIKE_BLOCKED ?
On 30 November 2011 12:20, Razvan Crainea wrote:
> Hi, Pete!
Hi, Pete!
Yes, indeed the RabbitMQ currently uses the same value for the exchange
name and routing key. Also only the '/' vhost can be used for the
moment, as it is hardcoded in the code. But this module is still under
development and any new ideas are welcome.
Regards,
--
Ra(zvan Crainea
O
Hi
I have a question about the RabbitMQ functionality.
The documentation states that the 'exchange' will also be used as the
'queue' used in RabbitMQ.
In my experience you do not send messages directly to a queue in RabbitMQ,
you send message which contains
- exchange name
- routing key
- messa
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