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

[Nhadie Ramos]

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 <ddunkin at netos.net> wrote:
From: Darryl Dunkin <ddunkin at netos.net>
Subject: RE: [asterisk-users] sometimes extensions can't be called
To: nhadie.ramos at yahoo.com, asterisk-users at lists.digium.com
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nhadie
Ramos

Sent: Tuesday, July 22, 2008 21:52

To: asterisk-users at lists.digium.com

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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