Hi All,
I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on
both asterisk. users register via domain, i have that domain on round-robin.
users can register and sometimes can call each other, but sometimes even if an
extension is register and i tried calling it, i got this on the the cli:
[Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
[Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1)
but xlite or ip phone shows the extension is registered. but asterisk says it's
busy. phones are behind NAT and using stun server. sip keep-alive is enabled
onxlite or ip phone. but it's just very inconsistent. i don't know where to
look at to fix this. any idea?
nhadie
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