[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                [EMAIL PROTECTED]