Hi, i see my extensions are there:
118103/118103 210.212.213.214 D N 5060 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118103/118103 210.212.213.214 D N 5060 Unmonitored and i have this on both servers: 17 sip peers [Monitored: 0 online, 0 offline Unmonitored: 15 online, 2 offline] regards, nhadie --- On Wed, 7/23/08, Darryl Dunkin <[EMAIL PROTECTED]> wrote: From: Darryl Dunkin <[EMAIL PROTECTED]> Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], [email protected] Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: [email protected] Subject: [asterisk-users] sometimes extensions can't be called 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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