title: How to connect asterisk to skype using siptosis?
body: <p>I've been trying to do this for weeks and still have not yet figured out how to connect to skype. The best progress I've found out was from this tutorial </p>
  
  <p><a href="http://translate.google.com/translate?hl=en&amp;sl=it&amp;u=http://www.voipandhack.it/archives/linux/asterisk-failover-e-registrazioni-sip&amp;ei=nqJRTNTPB8OfrAfDovGDAw&amp;sa=X&amp;oi=translate&amp;ct=result&amp;resnum=2&amp;ved=0CBoQ7gEwAQ&amp;prev=/search%3Fq%3Dasterisk%2Bauto%2Bfallthrough%2Bsiptosis%26hl%3Den" rel="nofollow noreferrer">http://translate.google.com/translate?hl=en&amp;sl=it&amp;u=http://www.voipandhack.it/archives/linux/asterisk-failover-e-registrazioni-sip&amp;ei=nqJRTNTPB8OfrAfDovGDAw&amp;sa=X&amp;oi=translate&amp;ct=result&amp;resnum=2&amp;ved=0CBoQ7gEwAQ&amp;prev=/search%3Fq%3Dasterisk%2Bauto%2Bfallthrough%2Bsiptosis%26hl%3Den</a></p>
  
  <p>But whenever I tried to make an echo123 call, my asterisk would show</p>
  
  <p>== Using SIP RTP CoS mark 5
      -- Executing [*123@phones:1] Dial("SIP/1004-00000030", "SIP/siptosisuser/echo123") in new stack
    == Using SIP RTP CoS mark 5
      -- Called siptosisuser/echo123
      -- SIP/siptosisuser-00000031 is ringing
      -- SIP/siptosisuser-00000031 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
      -- Auto fallthrough, channel 'SIP/1004-00000030' status is 'CONGESTION'</p>
  
  <p>and my siptosis will show:</p>
  
  <p>2010-07-30 10:48:21,596 Failed to select RTP format
  2010-07-30 10:48:21,597 ### local descriptor=v=0
  o=skypests 1280512101 0 IN IP4 192.168.<em>.</em>
  s=Session SIP/SDP
  c=IN IP4 192.168.<em>.</em>
  t=0 0
  m=audio 63202 RTP/AVP 0 8 98 101
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000
  a=rtpmap:98 iLBC/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15
  a=sendrecv
  a=silenceSupp:off</p>
  
  <p>2010-07-30 10:48:21,597 ### remote descriptor=v=0
  o=root 1729829715 1729829715 IN IP4 127.0.0.1
  s=Asterisk PBX 1.6.2.8
  c=IN IP4 127.0.0.1
  t=0 0
  m=audio 18104 RTP/AVP 3 101
  a=rtpmap:3 GSM/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-16
  a=silenceSupp:off - - - -
  a=ptime:20
  a=sendrecv</p>
  
  <p>I've changed my IP address to 192.168.<em>.</em> just for security reasons. It seems to be connecting one way or another but I'm assuming I'm doing something wrong somewhere in the code? I also haven't touched the siptosis.cfg because the tutorial I specified didn't do it. The cfg file also didn't come with the siptosis package I downloaded but I was able to find a copy from another tutorial on the nerdvittle website</p>
  
  <p>This is suppose to be the tutorial that makes it easy to setup but it doesn't seem to work on mine. I just want to have some call to echo123 and it seemed to be not working. I'm working on ubuntu and most tutorials are on CentOS so there's not much to solve my problem. I also am not using static skype, will that be a problem?</p>
  
  <p>Any hints/tip/answers would be very much appreciated! </p>
  
  <p>Thank you for your time and thank you in advance!</p>