[Asterisk-Users] SIP Trunking

[Steven Haldeman]

Thank you for your response.
   
  All that I get when I dial in is all circutes are busy and when I dial out 503 errors.  Here are my configs.  Any ideas would be greatly appreciated.  The provider is using a Tekelec 9000 Class 5 switch if that is any help.
   
  The provider sat us up two accounts one that they are calling a line that uses authentication usernames and passwords as though we were terminating into a SIP phone.  This account works but is not the prefered solution to our problem.  The other account is what the provider calls a trunk account and it does not use usernames and passwords.  This seems to be the solution for the provider and us.
   
   
  Thank you,
  Steven
   
   
  sip.conf
   
  [inbound-trunk]
type=friend
context=incoming
insecure=very
host=xxx.xxx.xxx.xxx
outboundproxy=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
nat=yes
canreinvite=yes
qualify=yes

  extension.conf
   
  [incoming]
exten => NXXNXXXXXX,1,Answer()
exten => NXXNXXXXXX,2,Background(greeting)
exten => NXXNXXXXXX,3,SayDigits(${CALLERIDNUM})
exten => NXXNXXXXXX,4,Dial(SIP/steven)
exten => NXXNXXXXXX,5,Hangup()


C F <shmaltz at gmail.com> wrote:
  It should work as is, just make usre that you have an extension
defined (or a catch all) for every DID you have with the provider so
that incoming works.

On 6/2/06, Steven Haldeman wrote:
>
> Hello,
>
> I am attempting to figure out how to set up SIP trunking, between my company
> and our SIP provider. This is an expermintal project at this time. The SIP
> provider gave us a Signalling IP address and two Media IP addresses. We
> supplied them with the IP address of our Asterisk box. When asked what our
> Usernames and Passwords would be we were told that they were not needed for
> a SIP trunk. We can use what they call SIP lines that use username/password
> however because of tarrifing the lines cost more per month than a trunk I
> have been successfull in making a SIP Line work, but have no idea where to
> start with a SIP Trunk. We sill be using DID numbers on the SIP trunks.
> Has anyone had any experiance with this type of configuration an example
> would be extrememly helpfull.
>
> I have search the internet for help and I may have seen a solution but just
> was not certain what I was looking at, or how to implement as everything
> that I have seen user a Username/Password combo.
>
> Thank you in advance.
>
> Thanks
>
> Steven
>
>
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