On Wed, Oct 31, 2012 at 10:31 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
I'm running Asterisk 10.7.0 with three sip trunks to my call termination
provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or
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
connection as it does not support non-SASL authentication. However, if
SASL is enabled, Asterisk
more specific, Asterisk Manager Interface , originate action will help you to
do so.
On Nov 9, 2012, at 5:34 PM, Danny Nicholas da...@debsinc.com wrote:
If you want to have a good level of control, AMI is the way to go. If you
just want simple and quick, .call files is going to do it. I
Hi,
I suppose WebRTC is the best solution nowadays, it's extremely interesting.
I developed a C2C app in 2008, starting with call files and AMI, ended
with asterisk-java and asterisk.NET to solve it.
Hint: Try to solve (al)most (all) of your problems using
Dialplans/Variables. Basically it's
Hi Marcus,
You're right,WebRTC is the way to go. The only drawback is the fact that
only astersik 11 support it natively.
On Sat, Nov 10, 2012 at 3:41 PM, Markus Weiler
markus_wei...@mailworks.orgwrote:
Hi,
I suppose WebRTC is the best solution nowadays, it's extremely interesting.
I
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 natively.
It's also not yet finished. Specifications are still being discussed,
finalized, completed. Implementations have certainly come a long way but
