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 a
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
Hi ,
l useing 0.6 version ,it`s working now ! thx
Best Regards,
yin
hualong@busap.com
From: Vlad Paiu
Date: 2013-09-20 00:15
To: users
Subject: Re: [OpenSIPS-Users] About install opensips include mongodb
Hello,
What mongo-c-client version are you using ? The OpenSIPS cachedb_mongodb
Hello!
I am using rtpproxy with Opensips for SIP connections.
Now rtpproxy makes a log to syslog during call establishment.
The question is may I shut off this process for rtpproxy working without
logging?
Thank you from any help.
I understand, thank you
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of kamika
Sent: Wednesday, September 25, 2013 3:48 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] rtpproxy logging shut off
you have to
You should decided if call should be answered or not BEFORE
forwarding it to asterisk, OR do this decision on asterisk side. Instead
stopping it at answer event.
Anyways, there a way to hangup running
calls (that are already established with 200 OK + ACK) using opensips
management interface,
Somebody have at least an idea if this is suposed to work ?
the package come to me, i check the did, change the $ru, and send to the
location, the location find the user and the package dont reach the other
side.
Thanks
2013/9/22 Mike Tesliuk m...@ultra.net.br
Hello Guys,
Im trying to
Hello,
The approach you're taking seems good, and it should definitely work.
Do you receive any errors in the OpenSIPS logs ? If you make a SIP trace
( ngrep / tcpdump ) on the OpenSIPS machine, do you see the INVITE
message getting out of OpenSIPS ?
If you don't see the package being sent
*OpenSIPS Summit, 11th of October 2013, Atlanta, US, collocated with
AstriCon Conference
*
This Summit is dedicated to OpenSIPS and Asterisk integration, focus
more on presenting technical tutorials - full description for solutions
to certain challenges related to integration.
Also visit *booth
Hello
I made the change (-d INFO:LOG_LOCAL5) but log from rtpproxy is still in
syslog.1 file.
In config file of syslog there is no mentioned about local7 log_facility.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of kamika
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
you should configure the nathelper and rtpproxy, this should help in you
issue.
2013/9/26 Rajesh Babu rajesh.b...@goodcoresoft.com
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
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...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Mike
14 matches
Mail list logo