[asterisk-users] Sip registration Asterisk 1.8

[Danny Nicholas]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, October 08, 2012 12:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Sip registration Asterisk 1.8

Hello,
I have a local Asterisk server that keep loosing its registration to main
Asterisk server. The local asterisk server is on the local subnet, it acts
as a client with extension 808. 

Local server
Sip.conf
register => 808:password at as2.xxxxx.com
registertimeout=20
registerattempts=10


Main Asterisk Server sip.conf

[808]
type=friend
context=sip-phones
call-limit=99
callerid="child2" <808>
disallow=all
allow=ulaw
allow=alaw
username=808
secret=xxxxx
dtmfmode=rfc2833
host=dynamic
mailbox=808
nat=yes
qualify=yes
canreinvite=no

  == Extension Changed 800[sip-phones] new state Idle for Notify User 812
[Oct  8 09:48:37] NOTICE[12030]: chan_sip.c:26141 sip_poke_noanswer: Peer
'808' is now UNREACHABLE!  Last qualify: 1
  == Using SIP RTP CoS mark 5


- Executing [808 at sip-phones:1] Dial("SIP/815-000000d8", "SIP/808,20,t") in
new stack
[Oct  8 09:49:02] WARNING[12277]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/815-000000d8' status is 'CHANUNAVA


Any ideas? 

Thanks in Advance!


--
IIRC qualify=yes means you get 60 seconds;  try it with qualify=300.