By using this site you agree to the use of cookies for analytics, personalised content and ads. Read more

404 not found when trying to make outbound calls

Novel Adventurer



Just got skype for business, incoming calls working fine but cannot get outbound to work.


Any ideas?


     Call-ID: YzUwYWZiNDRlMTc0NzhlNDlmYjExMmRjNWFmZDE3M2Q.
     CSeq: 1 INVITE
     Session-Expires: 1800
     Accept: application/sdp, message/sipfrag
     Content-Type: application/sdp
     Supported: replaces, histinfo, 100rel, timer
     User-Agent: TeleVoice Trio Enterprise 3.2 TE 3.2.79
     History-Info: <s
     o=- ****************330 1 IN IP4
     c=IN IP4
     t=0 0
     m=audio 53018 RTP/AVP 8 101 0
     a=rtpmap:8 pcma/8000
     a=rtpmap:101 telephone-event/8000
     a=rtpmap:0 pcmu/8000
     sigcomp id=
 Resiprocate: (level=7)
     Adding message to tx buffer to: [ V4  UDP target domain= mFlowKey=0 ]
 Resiprocate: (level=7)
     incoming from: [ V4  UDP target domain=unspecified mFlowKey=352 ]
 Resiprocate: (level=8)
     CSeq: 1 INVITE
     Content-Length: 0
 Resiprocate: (level=7)
     incoming from: [ V4 UDP target domain=unspecified mFlowKey=352 ]
 Resiprocate: (level=8)
     CSeq: 1 INVITE
     Content-Length: 0

 [edited for privacy]


See more topics labeled with:

Message 1 of 5 (2,000 Views)
4 Replies
Novel Adventurer

Problem solved, got some assistance from an asterisk guru...


Message 2 of 5 (1,941 Views)
Novel Adventurer

having the same issue using freeswitch, can't dial any phone number or another skype username.  have sufficient calling credits.

can you share your solution?

Message 3 of 5 (1,737 Views)
Novel Adventurer

can anyone over at Skype help, there seems to be no online chat support anymore, and I am a paying customer, very dissapointing!

I have Skype Connect  (skype sip) setup for incoming calls and this has been working well for some time, but now i want to call out to skype users using the same account and I get both 407 and 404 errors, a redacted trace copied below:


|Time     |                         |
|         |                   |    |                   
|0.000    |         INVITE SDP (g711U g711A telephone-eventRTPType...1 CN)          |SIP From: "" < To:<
|         |(5080)   ------------------>  (5060)   |
|0.041    |         100 Trying|                   |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.184    |         407 Proxy Authentication Required          |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.184    |         ACK       |                   |SIP Request
|         |(5080)   ------------------>  (5060)   |
|0.185    |         INVITE SDP (g711U g711A telephone-eventRTPType...1 CN)          |SIP From: "" < To:<
|         |(5080)   ------------------>  (5060)   |
|0.225    |         100 Trying|                   |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.857    |         404 Not Found                 |SIP Status
|         |(5080)   <------------------  (5060)   |
|0.857    |         ACK       |                   |SIP Request
|         |(5080)   ------------------>  (5060)   |

the (redacted) gateway config in freeswitch is:


        <gateway name="skype">
        <param name="username" value="9xxxxxxxxxxxxx"/>
        <param name="realm" value=""/>
         <param name="from-user" value="9xxxxxxxxxxxxx"/>
        <param name="from-domain" value=""/>
        <param name="password" value="XXXXXXXXXXXXXX"/>
         <param name="extension" value="9xxxxxxxxxxxxx"/>
        <param name="proxy" value=""/>
        <!-- <param name="register-proxy" value=""/> -->
        <param name="register" value="true"/>
  <!-- which transport to use for register -->
        <param name="register-transport" value="udp"/>
  <!--How many seconds before a retry when a failure or timeout occurs -->
        <param name="retry-seconds" value="30"/>
        <param name="caller-id-in-from" value="false"/>
        <param name="contact-params" value=""/>
        <param name="context" value="public"/>


Can someone over at skype support assist? or anyone else who has experience with this?
 The 404 appears to be the more serious error but not sure if the 407 is causing the 404 and needs to be cleared up first.


Message 4 of 5 (1,666 Views)
Novel Adventurer

ok, after playing around with this some more I was able to call me mobile, so my config is correct and working, although I do still see the 407 message, the 2nd invite seems to complete the call.  The wiki page on freeswitch says that dialing skype users directly does not work but the page is several years old so I'm dissappointed that this is still the case, even dialing my skype number in e.164 format fails with a 404 not found.


so my question to Skype is this:  Why can't we call out to skype users directly over skype for sip (skype connect), or even dial a user's skype phone number?  When will this be fixed?


Considering this is a pay for service it is really just 1/2 a service, and I need the other half to be fully working !


Message 5 of 5 (1,662 Views)
Discussion Stats
  • 4 replies
  • 0 kudos
  • 2 in conversation