[Asterisk-Users] sip to sip calls thru asterisk

Lyle Giese lyle at lcrcomputer.net
Thu Aug 26 17:10:17 MST 2004


You may need a stun server.  I setup my own and it appears to be working just fine.
http://www.vovida.org/applications/downloads/stun/

To * both sip clients appear to be on the same IP and that confuses *.  STUN should clear up some of that confusion.

I don't know enough about STUN to know if it needs to be on a public IP vs a private ip however.

Lyle

  ----- Original Message ----- 
  From: Gary Carr 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, August 24, 2004 1:29 PM
  Subject: [Asterisk-Users] sip to sip calls thru asterisk


  I have a test box setup and I can make outbound calls on the PSTN thru the diguim card, however I can not make a sip user to sip user call by dialing the extensions. I am getting the following error.

  -- Called cisco7960
      -- Got SIP response 482 "Loop Detected" back from 208.218.14.123
    == No one is available to answer at this time



  CLI> sip show peers
  Name/username    Host            Dyn Nat ACL Mask             Port     Status

  cisco7960/5052   208.218.14.123   D   N      255.255.255.255  5060     OK (1 ms)
  garycarr/5011    208.218.14.123   D   N      255.255.255.255  5060     OK (1 ms)


  sip.conf statements

  register => garycarr at sip.carolina.net/5011
  register => cisco7960 at sip.carolina.net/5052

  [cisco7960]
  type=friend
  host=dynamic
  nat=yes
  qualify=200
  dtmfmode=rfc2833
  canreinvite=no
  mailbox=5052
  callerid="Cisco 7960"
  context=local

  [garycarr]
  type=friend
  host=dynamic
  nat=yes
  qualify=200
  dtmfmode=rfc2833
  canreinvite=no
  mailbox=5011
  callerid="Gary Carr"
  context=local

  extensions.conf statements

  exten => 5011,1,dial(SIP/garycarr,20,tr)
  exten => 5052,1,dial(SIP/cisco7960,20,tr)

  Is this a possible nat issue? I can make a good call from behind the firewall doing sip to pstn so it seems 2 way traffic thru the firewall is working.


  I am still sifting thru the sip debug info but anyone has any ideas that would be great.


  Gary



------------------------------------------------------------------------------


  _______________________________________________
  Asterisk-Users mailing list
  Asterisk-Users at lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
     http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040826/4b9abaf6/attachment.htm


More information about the asterisk-users mailing list