[asterisk-users] Call is not coming through sipgate.co.uk+Asterisk

[Dovid B]

Are you behind NAT. Any firewall's ?
  ----- Original Message ----- 
  From: Crazy Boy 
  To: asterisk-users at lists.digium.com 
  Sent: Wednesday, October 25, 2006 10:54 AM
  Subject: [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk


  Hi,

  I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx. 
  I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I made a call, Asterisk console is also not showing anything. But, sipgate website is showing my calls list. I thought that When I made a call from outside to my number, call is going to sipgate.co.uk and its not routing to my server. When I execute "sip show registry", its not displaying anything. 

  Here I am giving my configuration details:

  My sip.conf file contents:

  [general]
  port = 5060
  bindaddr = 0.0.0.0
  qualify=no
  disable=all
  allow=alaw
  allow=alaw
  allow=ulaw
  allow=g729
  allow=gsm
  allow=slinear
  srvlookup=yes

  [250]
  type=friend
  username=250
  secret=danny
  callerid="Danny"
  host=dynamic
  context=demo

  register => 100xxxx:password at sipgate.co.uk/100xxxx

  [sipgate4]
  type=friend
  disallow=all
  allow=alaw
  allow=ulaw
  fromuser=100xxxx
  authuser=100xxxx
  secret=password
  username=100xxxx
  host=sipgate.co.uk
  context=demo
  dtmfmode=info
  fromdomain=sipgate.co.uk
  insecure=very
  nat=yes
  canreinvite=no
  callerid="Danny" <0207100xxxx>

  My Extensions.conf file contents:

  [demo]
  exten => 250,1,Dial(SIP/250,20)
  exten => 250,2,Voicemail(u250)
  exten => 250,3,Voicemail(b250)
  exten => 250,4,Hangup

  exten => _0207.,1,SetCallerID("" <100xxxx>|a)                ;Outgoing
  exten => _0207.,2,Dial(SIP/${EXTEN:4}@sipgate4,40,tr)

  exten => 100xxxx,1,Dial(SIP/250,30,tr)                               ;Incoming

  Am I have to install any other libraries?
  Anything wrong in the above configuration?

  Looking forward to your response. Thanks in advance.

  Regards,
  Chandra.



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