[Asterisk-Users] SIP Trunking

[Steven Haldeman]

I also forgot to mention that our provider gave use three different IP addresses.  One IP address for "Signalling" and two addresses for "Media."  If this helps any.
   
  Thank you,
  Steven

Steven Haldeman <haldeman_79 at yahoo.com> wrote:
    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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