Posted: Sun Apr 30, 2006 3:36 pm Post subject: [Asterisk-Users] Intermittent problem dialling out on a SIP
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:
Quote:
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
Quote:
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:
Quote:
-- 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
Quote:
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:
Quote:
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
;
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