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

[Stephen Foster]

Hi all,

 Im
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 IPs.



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 dont 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
havent 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 cant
figure out why that worked for them but I cant 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

[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.



Id 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/14Zap/7Zap/8Zap/9Zap/10Zap/11Zap/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]