[Asterisk-Users] Outgoing calls to SIP provider

[mtm spm]

Hello guys,

Software: Asterisk From CVS, Linux RH9
No telephony hardware.
I have a SIP service provider (addaline.com) and
(at this time) one Windows SIP (soft)phone.

Problem:
Both from console and from the sip phone if I dial
an autside number it doesn't work.

With sip debug I seen that addaline.com answer
with a (401 Unauthorized) message when Asterisk issue
a INVITE without authentification. However, after
receiving the 401 Asterisk doesn't send back another
INVITE with credentials. Why this happen?
When I call from the outside my number at addaline,
asterisk answer and send me to my soft phone without
any problem.

My settup is something like this:
sip.conf
  
[proxy.addaline.com]
type=friend
host=216.87.144.203
username=5555555555
secret=secret
context=from-sip

[2203]
type=friend
username=2203
secret=secret
host=xx.xx.xx.xx
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833


extension.conf:
[from-sip]
exten => _.,1,Answer
exten => _,1,Dial(SIP/2203,20)
exten => _,2,Hangup
exten => 2203,2,Voicemail(u2203)

[intern]
exten => _7.,1,Dial,SIP/proxy.addaline.com,20,tr

[local]
;include => from-sip
include => intern

Is there a problem in my setup or is something with
Asterisk?

Thank you,
MTM


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