Dial a sip trunk

Hi this is a simple question,
trying to Dial a SIP peer but when the peer prompts for authentication with 401 or 407, Asterisk doesn’t send any authentication and just fails the call. Does this look OK?

sip.conf :
[sipprovider]
type=peer
secret=password
host=1.1.1.1
fromuser=user
fromdomain=example.com
nat=yes
qualify=no

extensions.conf :
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipprovider)

After SIP provider reponse with 401, Asterisk should resending the Invite packet with MD5 encrypted authorized info.

Here is what work for me for my SIP trunk setting.
[TieUs SIP Trunk]
context=from-pstn
fromdomain=209.139.xxx.xxx
fromuser=604xxxxxxx
host=209.139.xxx.xxx
insecure=port,invite
secret=xxxxx
type=peer
defaultuser=604xxxxxxx

Also, have you pre-register the SIP trunk? because if you have the registration string already working, you should already see the SIP trunk regist properly in the Asterisk

Dan
TieUs Technology

I can’t see what is wrong with the original configuration, for outgoing calls, but in the proposed configuration, the use of insecure=port,invite is usually wrong. Most people only need insecure=invite, if they need anything. Service providers prefer the least secure setting becauase they have to think less.