[Asterisk-Users] Route SIP calls to provider

[Alexander Scheerschmidt]

Hi,
I hope this will help. I will give you my configuration. I'm using nikotel
to make international calls thru
My SIP provider.
Sip.conf 
 
[general]
port=5060
bindaddr=10.0.0.10
disallow=all
allow=g726
allow=alaw
reinvite=no
register => myid:mypasswd at calamar0.nikotel.com/myid

[2000]
type=friend
username=2000
secret=test
host=dynamic
context=from-sip
mailbox=100
disallow=all
allow=g729               ; Using g729, to get better sound quality on my
Zyxel P2000W
dtmfmode=rfc2833

[2001]
type=friend
username=2001
secret=test
host=dynamic
context=from-sip
mailbox=101
disallow=all
allow=g729                ; Using g729, to get better sound quality on my
Zyxel P2000W
dtmfmode=rfc2833

[2002]
type=friend
username=2002
secret=test
host=dynamic
context=from-sip
mailbox=102
dtmfmode=rfc2833         ; I'm not using g729 here, because I have a
excelent voice quality with the alaw or g726 codec
                         ; This is a wired IP phone 
[nikotel]
type=friend
context=sip-dial
username=myid
secret=mypasswd
host=calamar0.nikotel.com
fromuser=myid
fromdomain=nikotel.com
canreinvite=no
qualify=yes
disallow=all
allow=gsm
insecure=very
nat=yes
dtmfmode=info
tos=0x18

extentions.conf :
[general]
static=yes
writeprotect=yes

;Local SIP phones
[from-sip]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup

exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup

exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup

exten => 2003,1,Dial(SIP/2003,20)
exten => 2003,2,Voicemail(u2003)
exten => 2003,102,Voicemail(b2003)
exten => 2003,103,Hangup

;Call to Hell
exten => 666,1,Answer
exten => 666,2,MP3Player(/home/asterisk/satani.mp3)
exten => 666,202,Hangup

;Mailbox 
exten => 5555,1,VoiceMailMain2(s${CALLERIDNUM})
exten => 5555,2,Hangup

;Request your phonenumber
exten = 5595,1,Answer
exten = 5595,2,SayNumber(${CALLERIDNUM})
exten = 5595,3,Hangup

;Outgoing calls thru SIP Provider
exten => _00.,1,SetCallerID(004935179799272)
exten => _00.,2,Dial(SIP/${EXTEN:2}@nikotel)
exten => _00.,3,Hangup

;Incoming calls thru SIP Provider
[sip-dial]
exten => myid,1,Dial(SIP/2000&SIP/2001&SIP/2002)
exten => myid,102,Hangup

1.This works for incoming and outgoing calls thru your SIP provider. It was
hard working to get it fixed, but 
  Now I have no probs anymore. 
2.Why g729 : I bought a few licenses, to use with my Zyxel phones (better
voice quality). It's not required,
  but it's usefull to use with your sip phones, almost not required in your
SIP provider part.
3.Oh yeah, to make outgoing calls thru your SIP provider you just have to
dial 00 + phonenumber, ie.
  To make a call to germany : 004912321212. (but you can change this as
required)

Cheers,
Alexander Scheerschmidt, TnmT Org.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of iMRAN
Sent: Wednesday, 20 April 2005 18:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Route SIP calls to provider

Dear Pros,

Can anyone be kind enough to guide me to route calls to my SIP carrier.

I have configured * to as local PBX from Softphones to hardphones and vice
versa, the hardphone i have is AudioCodec MP108 8 FXS port gateway.

SIP.conf

[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729

[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833

[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833

extension.conf

[general]
static=yes
writeprotect=yes

[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000

[international]
ignorepat => 88
exten=> _1N1NXXNXXXXXX,1,Dial ???????

[internal]
include => local-sip

[local-sip]
exten => 1000,1,Dial(${PHONE1},40,t)
exten => 1000,2,Hangup

exten => 2000,1,Dial(${PHONE2},40,t)
exten => 2000,2,Hangup

exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup

i want user to dial 88 and they will get a tone and dial US or UK number
from local-sip context.

the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International context
in extension.conf also what i shld do in sip.conf..

my audiocodec supports g729 and g723 codec so do i need to aqquire license
for G729 from digium, if yes then why?

last if possible can you also please tell me wht i need to add on my context
so user can while in calling put the call on hold and transfer to another
sip phone.

I thankyou all for reading this mail and i hope someone will be kind enough
to help.

Best regards,

Imran
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