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

[Nhadie Ramos]

Hi,

I think i notice the problem now, but unfortunately i don't know how to fix it.

i'm using 118103 i dial 113102 i got this on asterisk server #1.

[Jul 23 18:27:48]     -- Called 118102
[Jul 23 18:27:49]     -- SIP/118102-08237ef0 is ringing

what i did is keep on dialing then hang up dial then  hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this:

[Jul 23 18:30:40]     -- Called 118102

but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk  #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see:

[Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2

yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit?

Thanks in advanced

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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