[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
=> _91NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1})

exten
=> _91NXXNXX,2,Congestion

 

[trunklocal]

exten
=> _9NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1})

exten
=> _9NXXNXX,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]

 

[Asterisk-Users] Using MultiTech MVP-210 as FXO/FXS gateway

[Stephen Foster]

Hi all,

    I’m
trying to use my 2-port multi-tech VoIP gateway to
talk to asterisk. Ideally I want to put it in a remote location with a POTS
line one port1 and an analog phone on port2 to call that location. Both the MultiTech and Asterisk have non-natted
static IP’s.

 

I have tried every different type of configuration possible
for the sip.conf file. I can call from the analog
phone on the multitech to a local asterisk extension
and it rings, but when I 
pickup I get a busy signal at both ends.

 

When I try and call from asterisk to the phone on the multitech, I don’t even get that far. I receive this
from the CLI:

 

    --
Starting simple switch on 'Zap/10-1'

    --
Executing Dial("Zap/10-1", "SIP/multitech") in new stack

    --
Called multitech

    --
Got SIP response 486 "Busy Here" back from 122.33.44.55

    --
SIP/multitech-964c is busy

  == Everyone is busy at this time

n  
Hungup 'Zap/10-1'

 

The MultiTech seems pretty simple
to configure, just the IP of asterisk, username and pass. The only field I
haven’t tried its SIP URL. I was recently at a MultiTech
show and I saw them use x-lite to call to the MultiTech. Since neither is a sip proxy, I can’t
figure out why that worked for them but I can’t get this working with
asterisk.

 

Here is the current version of my sip.conf

 

[multitech]

context=local

;disallow=all

allow=all

;disallow=all

allow=gsm

allow=ulaw

allow=alaw

type=friend

username=multitech

secret=pass

nat=no

;mailbox=200

host=dynamic

reinvite=no

;canreinvite=yes

qualify=1000

dtmfmode=info

canreinvite=no

callerid="Multi
Tech"

;defualtip=1.2.3.4

 

Thanks everyone,

    Steve