asterisk users - Apr 2006 - WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'

Hi,

Red Hat 9.0
Asterisk 1.2.7.1

Whenever I start Asterisk, I am unable to call out on my SIP channel:
>-- Executing Dial("Zap/1-1",
"SIP/6137451576@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)
However, if I RELOAD my dial plan from the CLI I get this message, it
starts to work.  I think I've tracked it down to the following warning
message:
>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable
to lookup '????'

This message doesn't reappear when I do a RELOAD.  Anyone know what
this is all about?  My SIP.conf is below.  Notice how I've commented
out auth=md5.  This seems to have eliminated the following WARNING
message that used appear just before the one above:
>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

Line 31 of my sip.conf was auth=md5.  I was able to get the SIP
channel working with this warning as well, but it took a lot more
RELOADs.

Any ideas?

SIP.conf
=====
[general]
;
context=incoming-bogus-calls
bindport=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
;
;---------------------------------------------------------------------

Hi,

Red Hat 9.0
Asterisk 1.2.7.1

Whenever I start Asterisk, I am unable to call out on my SIP channel:
>-- Executing Dial("Zap/1-1",
"SIP/6137451576@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)
However, if I RELOAD my dial plan from the CLI I get this message, it
starts to work.  I think I've tracked it down to the following warning
message:
>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable
to lookup '????'

This message doesn't reappear when I do a RELOAD.  Anyone know what
this is all about?  My SIP.conf is below.  Notice how I've commented
out auth=md5.  This seems to have eliminated the following WARNING
message that used appear just before the one above:
>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

Line 31 of my sip.conf was auth=md5.  I was able to get the SIP
channel working with this warning as well, but it took a lot more
RELOADs.

Any ideas on how I can get my SIP channel working properly, without
these warning messages & w/o having to do a RELOAD each time?

SIP.conf
=====
[general]
;
context=incoming-bogus-calls
bindport=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
;
;---------------------------------------------------------------------

Hi,

Red Hat 9.0
Asterisk 1.2.7.1

** Apologies if you notice this posted multiple times, I'm just not
seeing it on the boards  **

Whenever I start Asterisk, I am unable to call out on my SIP channel:
>-- Executing Dial("Zap/1-1",
"SIP/6137451576@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)
However, if I RELOAD my dial plan from the CLI I get this message, it
starts to work.  I think I've tracked it down to the following warning
message:
>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable
to lookup '????'

This message doesn't reappear when I do a RELOAD.  Anyone know what
this is all about?  My SIP.conf is below.  Notice how I've commented
out auth=md5.  This seems to have eliminated the following WARNING
message that used appear just before the one above:
>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

Line 31 of my sip.conf was auth=md5.  I was able to get the SIP
channel working with this warning as well, but it took a lot more
RELOADs.

Any ideas on how I can get my SIP channel working properly, without
these warning messages & w/o having to do a RELOAD each time?

SIP.conf
=====
[general]
;
context=incoming-bogus-calls
bindport=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
;
;---------------------------------------------------------------------