On Mon, Jan 10, 2011 at 8:09 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Jeff,
You can see how nat traversal is done with nathelper + RTPproxy - download
the opensips virtual machine
(http://www.voice-system.ro/shortcuts::opensips_livedvd) were you have a
ready to run opensips
Hi Sven,
Bogus Event 8 in state 2 is translated to receiving an indialog
request (non ACK, non BYE) while dialog in early state.maybe it is a
PRACK to the 183can you check that ?
Regards,
Bogdan
Sven Schulz wrote:
Running opensips 1.6.3, dialog module seems to function correctly
I'm trying to solve accounting + lines limit (numbers of incoming and
outgoing calls) + refer problem..
The problem seems as very hard to me.
Everything works well, but refer request.
There are 3 (A, B, C) UAs, each has limititation of number of incoming and
outgoing calls.
1. A calls B. Now A
Hi Jeff,
The UDP versus TCP issue is for the communication between opensips and
media relay ? If so, how comes you have a firewall between them ? you
have opensips and the media relay in different networks ?
Regards,
Bogdan
Jeff Chua wrote:
On Mon, Jan 10, 2011 at 8:09 PM, Bogdan-Andrei
Hi Bogdan,
Have you been able to take a look at the traces I sent?
Thanks
Diego
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Diego Barberio
Sent: lunes, 10 de enero de 2011 12:51 p.m.
To: users@lists.opensips.org
Hello Denis,
In the official release of RTPProxy, the timeout parameter (-T) controls
both session establishment and rtp timeout. This is a problem since we
would like to have a long period for call establishment, but a fast
media timeout detection.
In the patched version of RTPProxy, the -W
Hi Diego,
The bug seams to be in your callee device. Take a look at the 200 OK it
sends:
U 192.168.2.165:5061 - 192.168.2.165:5060
SIP/2.0 200 OK..From: Your Long
Namesip:usern...@192.168.2.150:5060;tag=A46E9878A6B36612423768382DD6C758..To:
I am having a problem with running opensips in debug level 6. When
opensips is set to this I am finding that it takes a long time to
respond to register requests (over 5 seconds compared to a fraction of a
second) which means that my phone times out when trying to register so I
cannot then
Razvan, I got rtpproxy from http://opensips.org/pub/rtpproxy/
http://opensips.org/pub/rtpproxy/ as you wrote.
I started it using such cli command “/usr/local/rtpproxy1/bin/rtpproxy -u
opensips -l 1.1.1.1 -s /var/run/rtpproxy.sock -T 80 -i -n /var/run/timer.sock
-d INFO” and made test call.
Hello Denis,
RTPProxy is only used to detect the media timeout. If OpenSIPS receives
a timeout notification on an unestablished call, it simply ignores it.
If you want to terminate the call when the callee doesn't answer you can
use the tm module and set the fr_inv_timer parameter. You can get
Hello, Razvan
“This is a problem since we would like to have a long period for call
establishment” and what does it mean “call establishment” in such context?
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday,
Hi Gareth,
Gareth Blades wrote:
I am having a problem with running opensips in debug level 6. When
opensips is set to this I am finding that it takes a long time to
respond to register requests (over 5 seconds compared to a fraction of
a second) which means that my phone times out when trying
Directly after the 183 Ringing is a NOTIFY message coming from the
destination (which is a Cisco Sip gateway). The source sends a corresponding
200 OK to this NOTIFY (also a cisco PBX).
So is the bogus event something I should be concerned with or is it more
of an informational error message?
Hello Denis,
A call is established when the callee answers it.
Regards,
Razvan
On 01/12/2011 04:52 PM, Denis Putyato wrote:
Hello, Razvan
“This is a problem since we would like to have a long period for call
establishment” and what does it mean “call establishment” in such context?
I installed a caching nameserver but it made no difference.
I then switched logging from syslog to stderr and thats much better and
the phone can register fine.
I have attached the debug from when I tried making the call between
lines. If there is nothing usefull there then can you let me know
Bogdan,
Wow, I didn't know about the live DVD.
Any chance someone could create this as an OpenVZ container in
addition to VMWare?
-- James
On Mon, Jan 10, 2011 at 2:25 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Damon,
Well, the answer is simple - download the opensips virtual
James, never user openVZ so far..there are a log of VM technologies out
there :)For the moment we release the opensips live distro on VMware
as that;s the main what we used...not sure what are the other main VM
tech used by other people...
Regards,
Bogdan
James Lamanna wrote:
Bogdan,
Hi Sven,
Interesting, never saw a NOTIFY in early state of a dialog...can you
post a SIP capture for such a dialog ?
Going back to your question, the message said that the notify event does
not fit to the current dialog state, but this has no effect on the
dialog state, neither on the
Jeff Chua wrote:
On Wed, Jan 12, 2011 at 7:56 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Jeff,
The UDP versus TCP issue is for the communication between opensips and media
relay ? If so, how comes you have a firewall between them ? you have
opensips and the media relay in
Hi Bogdan,
Thanks again for your response.
I understand your point but as fair as I know if the contact is set to
50257609...@192.168.2.165:5061 the subsequent messages (i.e. ACK and BYE)
will be sent directly to the callee bypassing the proxy. This is OK for me,
but I understand that the LB
That's not true - the contact address is the address of the other end
point, it does not mean that the communication is done directly between
the end points. The Route hdr is the one dictating the intermediary
hopsbottom line , contact points the end point, sequential requests
will visit
Hi Bogdan,
Thanks for your prompt response and all your support. I will correct that
bug, and test again.
Thanks
Diego
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: miércoles, 12 de enero de
On Thu, Jan 13, 2011 at 3:10 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
where is the nat part here ? can you directly route between OpenSIPS B and C
?
Bogdan,
I'm trying to use my iPhone from home to route via my PC (vpn). It's
more like this below ...
iPhone, wlan, home PC has
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