[Asterisk-Users] Intermittent problem dialling out on a SIP channel

[hugolivude]

Hi,

Red Hat 9.0
Asterisk 1.2.7.1

I'm having a bit of an intermittent problem with my SIP account. 
Often (but not always) when I start * or RELOAD my dial plan from the
CLI I get this message:

>Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822
add_realm_authentication: Format for >authentication entry is
user[:secret]@realm at line 31
>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable
to lookup '????'

Line 31 of my sip.conf is auth=md5
.  Whenever I see that message, I am unable to dial out on the SIP channel:

>-- Executing Dial("Zap/1-1", "SIP/6137451576 at 6477235412||t|") in new stack
>Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such
host: 6477235412
>Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create
>channel of type 'SIP' (cause 3 - No route to destination)

If I repeatedly RELOAD enough times from the CLI though, eventually
one will work without the error messages and I can dial out.

I tried commenting out auth=md5 in my SIP.conf.  That seemed to
eliminate the "add_realm_authentication" error, but I still see the
"ast_get_ip_or_srv" from time to time, and when I do, I can't dial
out.

Also, while I am successful at dialling out from time-to-time,
depending upon how the RELOAD goes, I havn't yet been able to receive
a SIP call.

Finally, another thing that troubles me is that sometimes I can use
QUIT or EXIT to exit the CLI, but other times it just doesn't work as
shown below:

>Use EXIT or QUIT to exit the asterisk console
> Reloading MGCP
>  == Parsing '/etc/asterisk/mgcp.conf': Found
>  == MGCP Listening on 0.0.0.0:2727
>  == Using TOS bits 0
>
>Use EXIT or QUIT to exit the asterisk console
>  == Parsing '/etc/asterisk/sip_notify.conf': Found
>*CLI>quit
>No such command 'quit' (type 'help' for help)
>*CLI> QUIT
>No such command 'QUIT' (type 'help' for help)
>*CLI> EXIT
>No such command 'EXIT' (type 'help' for help)

Any ideas?  My sip.conf is provided below:

[general]
;
context=incoming-bogus-calls
port=5060                 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0          ; Address to bind to (all addresses on machine)
maxexpirey=3600			  ; Must be larger than the re-register timeout on the router
defaultexpirey=3600
notifymimetype=text/plain
rtptimeout=60
rtpholdtimeout=300
disallow=all
allow=ulaw
;
; This section is because i'm behind nat
;

register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412
externip=<mystaticIPaddress> ;Outside address
localnet=192.168.0.148/255.255.255.0 ;Inside Network
;
;********************************************************************
[6477235412]
type=peer
auth=md5
username=6477235412
fromuser=6477235412
fromdomain=unlimitel.ca
secret=<mypassword>
host=sip.unlimitel.ca
port=5060
nat=yes
canreinvite=no
qualify=no
disallow=all
allow=g729
dtmfmode=rfc2833
insecure=very
context=incoming
;
;---------------------------------------------------------------------