On Thursday 24 Apr 2014, Mikael Fredin wrote:
I will look into netcat as well, thank you
There's not much to look into, really! It's just a command-line tool for
connecting STDIN and STDOUT to a network socket.
$ echo -e WIBBLE\nWIBBLE\nWIBBLE | nc somehost.co.uk 3245
will send
WIBBLE
WIBBLE
Is there a way to divert incoming calls on DAHDI T1 channels so telco gets
the diversion and send the call to new number and releasing the channel?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or
chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly,
will ast negotiate srtp or dtls even ast and the proxy speak sip in
the clear over the lo interface?
Avoiding encryption over lo can aid debugging, but will
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been
heavily modified. Currently asterisk runs on
El 25/04/14 18:29, Alex Villacís Lasso escribió:
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has
James Cloos wrote:
Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or
chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly,
will ast negotiate srtp or dtls even ast and the proxy speak sip in
the clear over the lo interface?
Avoiding encryption over lo can aid
