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

[Tim Robinson]

 From a brief look, it seems you do not have a context= in your sip.conf 
file for the extension.  If you don't put a contxt in, I don't know what 
it assumes, and it will not include the contexts you have set up to 
define external access.

 From looking at your dialplan, if you put context=local into the [gs01] 
entry in sip.conf, you should be able to make outbound calls from this 
extension, as you will be forced into 'local' context and will be able 
se see all the external access contexts you have defined.

Let us know how you get on...

Rgds
Tim Robinson
Basingstoke, UK

Stephen Foster wrote:
> 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
> 
>  
>