I would be interested to know how good or helpful their support is as I have 
shortlisted les.net as my provider too.
Feel free to let me know if they can resolve your problem.

Thanks,
Richard
  ----- Original Message ----- 
  From: Aloysius Thevarajah Lloyd 
  To: Dave Donovan 
  Cc: [email protected] 
  Sent: Tuesday, February 06, 2007 11:55 AM
  Subject: Re: [on-asterisk] termination via les.net


  why do not you send a email to les.net.

  He will help you to solve the problem.

  Thanks
  Lloyd

   
  On 2/6/07, Dave Donovan <[EMAIL PROTECTED]> wrote: 
    Hi David,

    Do you get an error message on the Asterisk console?  It's often helpful to 
see what's happening by turning on a verbose logging mode with "set verbose 20" 
or something like that.  Consider posting that here, it's often helpful in 
debugging. 

    Do you get audio in one direction?

    I haven't used Les.net but maybe if you are able to post some more info, 
someone on the list will be able to help.  Personally most of my "SIP 
connection problems" have boiled down to a typo in an IP address or the 
dialplan or something. 

    Dave 



    On 2/5/07, David Sfiligoi <[EMAIL PROTECTED] > wrote: 
      Hi,

      Based on an earlier thread where some taug users les.net  was pretty good 
I decided to give it a try. I have been banging my head against this problem 
for a while since probably it is my own problem.  So far no problem receiving 
phone call when configured on both SIP and IAX. However, I cannot seem to make 
outbound phone call work on SIP(it does woek on IAX). WhenI phone I get some 
lday saying that the peer is incorectly configured.  I have tried many 
things(including what les recommends), just can't seem to get it.  Using some 
other voip company on sip I have no issue on sip. Anybody has some sample 
config that works well?(please remove your personal info if you post your 
config) Alternatively anybody that can help. 

      Here the relevant sip.conf

      [lesnet_peer]
      type=friend
      host=did.voip.les.net
      canreinvite=no 
      dtmfmode=rfc2833
      insecure=very
      disallow=all
      allow=ulaw
      allow=gsm
      nat=yes
      username=peername
      secret=itssecret


      register => peername:[EMAIL PROTECTED]/peername

      and extensions.conf

      exten => _6NX.,1,Dial(SIP/lesnet_peer/${EXTEN:1})
      exten => _6NX.,2,Congestion
      exten => _61NX.,1,Dial(SIP/lesnet_peer/${EXTEN:1}) 
      exten => _61NX.,2,Congestion

      thanks,
      David

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