[asterisk-users] sometimes extensions can't be called

Nhadie Ramos nhadie.ramos at yahoo.com
Tue Jul 22 23:51:41 CDT 2008


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



      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/30bb8669/attachment.htm 


More information about the asterisk-users mailing list