Asterisk system help

agni_suresh896
Joined: Wed 06 of Dec, 2006

Re: Asterisk system help

Posted:Sat 10 of Jan, 2009 (20:02 UTC)
I would like to assist you(with out any cost). I am a developer from India. You can reach me on my mobile +91 9849517562 or through my email : agni.suresh896@gmail.com.
shang
Joined: Thu 18 of Dec, 2008

Re: Asterisk system help

Posted:Fri 09 of Jan, 2009 (00:08 UTC)
thanks for your reply halsp. Our PSTN lines are working now. At one time the sip trunk was working as well, but is no longer. That is the reason I sent the entire config, because I believe there may be an error somewhere in it that is causing it not to work. I don't know enough about Asterisk to catch such an error.

Also, I would like to have a developer on hand that would be able to connect into our system and make changes from time to time at whatever would be a fair price to do so.
halesp
Joined: Wed 15 of Mar, 2006

Re: Asterisk system help

Posted:Tue 06 of Jan, 2009 (04:19 UTC)
The Asterisk-users mailing list is good, but it's best to post one question at a time.
(not send your whole system config in)..And also work on one problem at a time - for example, get your incoming calls working on the PSTN lines first.
Failing that, you could contact a company that does Asterisk support and pay them to get it all working!
shang
Joined: Thu 18 of Dec, 2008

Re: Asterisk system help

Posted:Mon 05 of Jan, 2009 (10:10 UTC)
It has been 3 weeks and no answers to my post. I guess I am in the wrong place. Does anyone know of a better suited forum for help with Asterisks configuration?
shang
Joined: Thu 18 of Dec, 2008

Asterisk system help

Posted:Thu 18 of Dec, 2008 (06:03 UTC)
Recently I purchased a asterisk phone system in which the company I bought from would set up to our needs. After working a short time on my configuration files he told me he would be back the next day to finish. Long story short he never came back and I am unable to reach him any longer.

I have tried to get the system up and running myself but after 2 weeks I am unsuccessful and need help.

If kind person can take a look at my extensions.conf and sip.conf and tell me what is wrong with it and how could I change it to get it working.

I have 5 phones connected to 2 PSTN and 1 VOIP line. The PSTN lines are in Thailand and the VOIP is the US.

SIP.CONF

[general]
bindaddr=0.0.0.0
bindport=5060
port=5060
g726nonstandard=yes
videosupport=yes
localnet=192.168.1.0/255.255.255.0
;localnet=10.0.0.0/255.0.0.0
;localnet=172.16.0.0/12
;localnet=169.254.0.0/255.255.0.0
tos_audio=ef
vmexten=voicemail
srvlookup=yes
context=from-sip-user
tos_sip=cs3
allowoverlap=no
disallow=all
allow=g729,g723,ilbc,gsm,g726,g726aal2,g711d,g711o,h263p,h263,h261,ulaw
trustrpid=yes
sendrpid=yes
dtmfmode=inband
relaxdtmf=yes
realm=asterisk
recordhistory=yes

register => register info

[authentication]

[101]
type=friend
dtmfmode=auto
restrictcid=no
nat=no
mailbox=101@default
secret=password
username=101
context=outgoing
host=dynamic
callerid=<101>
allow=all

[201]
type=friend
host=dynamic
secret=password
username=201
context=outgoing
callerid=<201>
accountcode=GPG1
dtmfmode=auto
nat=no
allow=all

[202]
type=friend
nat=no
secret=password
host=dynamic
username=202
context=outgoing
callerid=<202>
allow=all

[301]
type=friend
nat=no
context=outgoing
restrictcid=no
dtmfmode=auto
secret=password
username=301
callerid=<301>
mailbox=301@default
host=dynamic
allow=all

[501]
type=friend
context=outgoing
nat=no
secret=password
username=501
amaflags=default
mailbox=501@default
callerid=<501>
host=dynamic
allow=all

[901]
type=friend
username=901
dtmfmode=auto
nat=no
secret=password
callerid="Office Fax" <901>
context=outgoing
host=dynamic
allow=all
qualify=yes
disallow=all
allow=ulaw
allow=alaw


; TRUNK CONFIGURATION
[viatalk]
type=friend
authuser=1972xxxxxxx
username=1972xxxxxxx
fromuser=1972xxxxxxx
fromdomain=domain.name.com
host=domain.name.com
context=viatalk_outgoing
secret=password
insecure=very
qualify=3600
nat=no ; switch to yes if behind nat (try to avoid it if at all possible)

; PEER CONFIGURATION
[1000] 
type=peer
nat=yes ; allows you to use a softphone/adapter behind nat
host=dynamic
canreinvite=yes
username=1000 
secret=password ; this can be anything you want 


EXTENSIONS.CONF

[general]
static=yes
writeprotect=yes

[globals]
CONSOLE=Console/dsp
NPX=214
PEER=1000 ; The peer you setup in sip.conf for your softphone/adapter
TRUNK=viatalk ; The name of the trunk you defined

[fxo1]
exten => s,1,Answer()
exten => s,2,Set(TIMEOUT(response)=2)
exten => s,3,Set(TIMEOUT(digit)=1)
exten => s,4,Background(welcome)

exten => t,1,Goto(fxo1,s,1)
exten => i,1,Goto(fxo1,s,1)
exten => 1,1,Goto(sales,s,1)		; sales
exten => 2,1,Macro(internal,301)		; Customer Service
exten => 3,1,Macro(internal,301)		; accounts
exten => 0,1,Goto(operator,s,1)		; operator
exten => 888,1,Goto(disaaccess,s,1)	; Out Line
exten => 9,1,Goto(fxo1english,s,1)		; LANGUAGE

include => internalextensions

[fxo1english]
exten => s,1,Set(TIMEOUT(response)=2)
exten => s,2,Set(TIMEOUT(digit)=2)
exten => s,3,Background(welcomeenglish)

exten => t,1,Goto(fxo1english)
exten => i,1,Goto(fxo1english)
exten => 1,1,Macro(internal,301)
exten => 2,1,Macro(internal,301)
exten => 3,1,Macro(internal,301)

include => internalextensions

[fxo2] ; fax
exten => s,1,Dial(SIP/901)

;[fxo3] ; us
;exten => s,1,Dial(SIP/501)

[outgoing]
include => internalextensions
include => viatalk_outgoing
include => thailines

[thailines]
exten => _X.,1,Dial(MSPD/thai/${EXTEN})

[incoming]
exten => _1X.,1,Dial(MSPD/us/${EXTEN:1})

[internalextensions]
exten => 101,1,Macro(internal,${EXTEN})
exten => _20[1-2],1,Macro(internal,${EXTEN})
exten => 301,1,Macro(internal,${EXTEN})
exten => 501,1,Macro(internal,${EXTEN})
;exten => 111,1,Record(welcome.wav)
;exten => 112,1,Record(welcomeenglish.wav)
exten => *,1,VoiceMailMain(${CALLERID(NUM)})
exten => **,1,VoiceMailMain()

[viatalk_incoming]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(response)=2)
exten => s,n,Set(TIMEOUT(digit)=1)
exten => s,n,Background(uswelcome)

exten => t,1,Goto(viatalk_incoming,s,1)
exten => i,1,Goto(viatalk_incoming,s,1)
exten => 1,1,Goto(sales,s,1)		; sales
exten => 2,1,Macro(internal,301)		; Customer Service
exten => 3,1,Macro(internal,301)		; accounts
exten => 0,1,Goto(operator,s,1)		; operator
exten => 888,1,Goto(disaaccess,s,1)	; Out Line

include => internalextensions

[viatalk_outgoing]
exten => 911,1,Dial(SIP/911@${TRUNK},60,r)
exten => 411,1,Dial(SIP/411@${TRUNK},60,r)
exten => *123,1,Dial(SIP/*123@${TRUNK},60,r)

exten => _NXXXXXX,1,Goto(1${NPX}${EXTEN},1) ; if dialing 7 digits, prepend 1 + Area Code
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) ; if dialing 10 digits, prepend 1

exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@${TRUNK},60,r)
exten => _1NXXNXXXXXX,2,Playtones(480+620/250,0/250) ; play a fast busy (reorder) tone
exten => _1NXXNXXXXXX,3,Congestion
exten => i,1,Hangup
exten => t,1,Hangup

[macro-internal]
exten => s,1,Dial(SIP/${ARG1},20)
exten => s,2,Voicemail(u${ARG1})

[sales]
exten => s,1,Dial(SIP/101,15)
exten => s,2,Dial(SIP/301,15)
exten => s,3,Voicemail(u101)

[operator]
exten => s,1,Dial(SIP/101&SIP/201&SIP/202&SIP/301,25)
exten => s,2,Voicemail(u301)

[disaaccess]

exten => s,1,Authenticate(/etc/asterisk/password.conf)
;exten => s,2,DISA(no-password|outgoing)
exten => s,2,Background(privacy-prompt)
exten => s,3,WaitExten(30)

exten => _X.,1,Dial(Local/${EXTEN}@outgoing)