[Asterisk-Users] Route SIP calls to provider

[Wiley Siler]

>From your descriptions of your needs, you would be better served with an
AAH installation.  Easier to understand than hand coding your contexts.

That aside, here are few answers...

Look here for more...
www.voip-info.org

 
Routing to the VoIP is just a matter of dial plan matching (see dial
plan at voip-info)

Codecs on Asterisk require the license.  Your phone may support the
codec but your server needs a license to do so.

Hold and transfer are usually part of the phone itself.  For example, my
Polycom holds the line and lets me transfer from itself.

Otherwise, I can transfer a call via * by dialing # and the extension
target.
This is a blind transfer so the target extension just gets the call
without you even talking to them.
Consultative transfer is different, details located at www.voip-info.org

Cheers,
Wiley



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of iMRAN
Sent: Wednesday, April 20, 2005 9:36 AM
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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