Hi All,
We are facing few problems, when device A register itself on OpenSIPS server
with timeout=600 in UDP transmission, after few seconds device disconnects
from internet due to any issue, internet unavailability or device shutdown.
Its registration maintain at server till it timeout.
Hi,
Try using a Linphone Client on Android device. I guess the connection issue
gets solved. Generally it gets Symmetric NAT formed on the server.
-Cheers
Rajesh
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ilnar Nigmatullin
Sent:
Hi
For debugging purpose use only linphone client on both sides. As linphone has
internal stun and ice configured to crack the NATing and by the way are you
using any media proxy or rtp proxy for handling the nating on the server
Sent by Maxis from my BlackBerry® smartphone
-Original
Hi All,
I am facing a issue with my messaging on opensips. The scenario are as
follows,
1. When UserAgent have some device failure or network connectivity down in
that case user presence still maintained, I can see the disconnected user
still online using ./opensipctil online, in
Hi All,
I use Opensips 1.9.1 and have enabled RTP and Nating in the configuration,
Whenever I use to connect the calls using my 3G connection, call gets
connected but my voice is not being heard, whereas though wifi everything is
working fine. I tried connecting with Linphone I didn't face
are using RTPProxy or
MediaProxy to handle media when originated from NATed clients. If yes, you
dont need STUN and TURN as of now.
--- Jayesh
On Fri, Mar 7, 2014 at 5:01 PM, Rajesh Babu rajesh.b...@goodcoresoft.com
wrote:
Hi All,
I use Opensips 1.9.1 and have enabled RTP and Nating
below,
http://pastebin.com/MPw8vYTf
Can anyone advice me whether should I expose the RTP Proxy and a NAT
traversal for messaging as well. As my server is not running on RTP or NAT.
Thanks Regards,
Rajesh Babu Vasudevan
___
Users
Hi All,
I have a requirement to support Groupchat on top of SIMPLE. Can anyone
throw some light on please?
Thanks in advance
-Rajesh
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
(just for test purpose)
you should increment you debug info too
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
2013/9/28 Rajesh Babu rajesh.b...@goodcoresoft.com
Hi,
I have attached the logs and my routing file @
http
knowledge on this. I am still under
learning stage please forgive me for asking some basic questions.
Thanks
Rajesh
-Original Message-
From: Aamir [mailto:aamir_...@yahoo.com]
Sent: Wednesday, 9 October, 2013 5:34 PM
To: Rajesh Babu
Subject: Re: [OpenSIPS-Users] OverSIP+Opensip
Let us
Hi,
After i installed OverSIP (Websocket implementation) by following the
tutorial mentioned @
http://www.opensips.org/Documentation/Tutorials-WebSocket. I am facing the
following issues,
1. My Audio and Video for Outside Network stopped working (Implemented
NAT helper RTPProxy),
and paste in some pastebin website and show us the link
2013/9/27 Rajesh Babu rajesh.b...@goodcoresoft.com
I am getting Error 483, too many Hops, There is no other error messages i am
getting. Please some one help me out in this
From: users-boun...@lists.opensips.org
[mailto:users-boun
, but i think that this is the start point :)
and if you is new to opensips i recommend to you the book about opensips (
http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book )
2013/9/26 Rajesh Babu rajesh.b...@goodcoresoft.com
Hi Mike,
Thanks for the response, I am
://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book )
2013/9/26 Rajesh Babu rajesh.b...@goodcoresoft.com
Hi Mike,
Thanks for the response, I am totally new to this world, can you please
help me by directing to on how to configure links. It will be great.
Thanks in advance
I am getting Error 483, too many Hops, There is no other error messages i am
getting. Please some one help me out in this
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Rajesh Babu
Sent: Friday, 27 September, 2013 6:08 PM
To: 'OpenSIPS users
/book )
2013/9/26 Rajesh Babu rajesh.b...@goodcoresoft.com
Hi Mike,
Thanks for the response, I am totally new to this world, can you please
help me by directing to on how to configure links. It will be great.
Thanks in advance
Regards
Rajesh
From: users-boun
Attaching the content
From: Rajesh Babu [mailto:rajesh.b...@goodcoresoft.com]
Sent: Wednesday, 25 September, 2013 3:27 PM
To: 'users@lists.opensips.org'
Subject: Audio and Video not working
Hi,
I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make
Hi,
I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make a Call inside the network between two users i don't have any
issue, where as from outside the network, even though i can see the user
registered in my server i am not able to call registered user (I see
From: Rajesh Babu [mailto:rajesh.b...@goodcoresoft.com]
Sent: Wednesday, 25 September, 2013 3:27 PM
To: 'users@lists.opensips.org'
Subject: Audio and Video not working
Hi,
I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make a Call inside the network
Hi,
I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make a Call inside the network between two users i don't have any
issue, where as from outside the network, even though i can see the user
registered in my server i am not able to call registered user (I see
Tesliuk
Sent: Thursday, 26 September, 2013 12:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
you should configure the nathelper and rtpproxy, this should help in you
issue.
2013/9/26 Rajesh Babu rajesh.b
21 matches
Mail list logo
