dtlsenable=yes was missing
thank you joshua
Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a):
hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with
HiI want to load balance SIP calls between two(or more)
Asterisks with only DNS SRV. I used bidirectional sync
Unison to synchronize configuration files and internal database file between
two Asterisk boxes.The problem is when a calls come to Asterisk1 but
SIPendpoint is registered on
I'm pretty sure there isn't a way to do that currently. My best guess
would be that a new special type of bridge technology could be created that
would implement the per-channel echo (no audio bridged between channels in
the bridge). That would require new C code in Asterisk for the bridge, and
Thanks for answer, AGI/AMI looks still rocks, will think about using ARI just
for queues and conferences.
Sent from my iPhone
On 25 May 2015, at 04:55, Scott Griepentrog sgriepent...@digium.com wrote:
I'm pretty sure there isn't a way to do that currently. My best guess would
be that a
