asterisk users - Jul 2008 - 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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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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