Hi all,
I'm trying to troubleshoot an issue with my SIP service. All outgoing
calls work normally. The following is a SIP debug log from Asterisk. The
test setup is as follows:
One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk
to my local FreePBX/Asterisk 11.0 server
It seems a firewall or signaling problem. The calling part is not
sending a ACK response to your host because it never get an OK 200
from your host.
In other words, the called part is trying to send to the calling part
a) TRYING 100, then
b) RING 180 and finally
c) OK 200
but the calling
On Sat, 10 Nov 2012 18:08:58 +
Phil Reynolds phil-aster...@tinsleyviaduct.com wrote:
I have tried today to change my Jabber server from OpenFire to
ejabberd. Unfortunately I have not been able to get Asterisk logged in
to Jabber since. If SASL is not enabled, then ejabberd rejects the
Am 11.11.2012 11:46, schrieb Eric Kuhnke:
I'm trying to troubleshoot an issue with my SIP service. All outgoing
calls work normally. The following is a SIP debug log from Asterisk. The
test setup is as follows:
Miguel already explained what's going on. Have a look at the SIP packets
to
On Mon, Nov 12, 2012 at 11:17 AM, Markus unive...@truemetal.org wrote:
Am 11.11.2012 11:46, schrieb Eric Kuhnke:
I'm trying to troubleshoot an issue with my SIP service. All outgoing
calls work normally. The following is a SIP debug log from Asterisk. The
test setup is as follows:
Hi all,
based on the following link, I am going to authenticate SIP asterisk users via
Radius client that is installed on my Asterisk then the radius client connect
to asterisk using the radius and ldap:
You can use Radius Agi developed by PortaOne from following link.
http://www.voip-info.org/wiki/view/PortaOne+Radius+auth
Regards,
Qasim
On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini samiramhosse...@yahoo.com
wrote:
Hi all,
based on the following link, I am going to authenticate SIP
Hi All,
I need to install and configuration of Aculab prosody X PCI card with
Asterisk-1.8.9.1 on Centos-5.7 system.
I will try for that but not success. so, please suggest me way to achieve
it.
Thanks in Advance.
--
Best Regards,
Rajni Vanza
--
its too early for webrtc. im also waiting for further development on this.
On Sun, Nov 11, 2012 at 3:51 AM, Joshua Colp jc...@digium.com wrote:
Adolphe Cher-Aime wrote:
Hi Marcus,
You're right,WebRTC is the way to go. The only drawback is the fact that
only astersik 11 support it
