[Asterisk-Users] Not able to dial "9" to get out with SIP Grandstream BudgeTone-100 or SIP softphone

[Stephen Foster]

Hi everyone,
                        I am having problems dialing "9" to get an
external line with my SIP phones or SIP clients. I have been looking for
months on websites, sitting in MIRC rooms, and reading * documentation
but I cannot seem to find a solution.
 
My asterisk box is sitting directly on the internet ( NO NAT ) with a
firewall. I have also tested this box on my LAN and I have the same
issue ( this is not a firewall issue ). I am using a T-100P card and an
Adtran Total Access unit for all my analog phones which for now is all I
use.
 
My Grand stream SIP phone works fine for calling internal extensions
with no problems at all. When I try and dial "9" and a number, after a
wait of a few seconds I get " 404 " displayed on the screen and a busy
signal. I have tried to tweak everything I know within the dial plan,
but I always seem to have the same issue. 
 
I previously tried to attach my sip and extensions.conf but the email is
too big for the mailing list. I have pasted small sections of them
below.
 
I'd very much appreciate any help anyone can provide.
 
SIP Conf
 
[gs01]
type=friend
username=gs01
secret=pass
nat=1
host=dynamic
qualify=yes
dtmfmode=info
canreinvite=no
 
EXTENSIONS.CONF
 
[general]
static=yes
writeprotect=no
 
[globals]
CONSOLE=Console/dsp
TRUNK=Zap/g2 RINGOUT=Zap/14&Zap/7&Zap/8&Zap/9&Zap/10&Zap/11&Zap/12
 
[trunkint]
exten => _9011.,1,Dial(${TRUNK}/www${EXTEN:1})
exten => _9011.,2,Congestion
 
[trunkld]
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/www${EXTEN:1})
exten => _91NXXNXXXXXX,2,Congestion
 
[trunklocal]
exten => _9NXXNXXXXXX,1,Dial(${TRUNK}/www${EXTEN:1})
exten => _9NXXNXXXXXX,2,Congestion
exten => 9411,1,Dial(${TRUNK}/www${EXTEN:1})
exten => 9411,2,Congestion
exten => 9911,1,Dial(${TRUNK}/www${EXTEN:1})
exten => 9911,2,Congestion
 
[local]
;trusted users only!
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => trunktollfree
include => trunkint
include => trunkld
include => phones
include => voicemail
include => recording
 
[macro-stdexten]
exten => s,1,Dial(${ARG2},20)
exten => s,2,Voicemail2(u${ARG1})
exten => s,3,Goto(default,s,1)
exten => s,102,Voicemail2(b${ARG1})
exten => s,103,Goto(default,s,1)
 
[phones]
exten => 200,1,Macro(stdexten,200,Zap/10)
 
;SIP phones
;Grandstream Phones
exten => 210,1,Dial(SIP/gs01)
exten => 222,1,Dial(SIP/bradwell)
exten => _64xx,1,Dial(SIP/gs${EXTEN:2}|20)
exten => _64xx,2,Voicemail2(u${ARG1})
exten => _64xx,3,Congestion
exten => _64xx,102,Voicemail2(b${ARG1})
exten => _64xx,103,Congestion
 
[sipstart]
include => phones
include => voicemail
include => default
include => trunklocal
include => trunktollfree
 
Thanks,
            Steve                sfoster at isense.ca
 
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