Skype for SIP has no buddy list support and no outgoing calling to Skype users, both of which Skype for Asterisk does have. Skype for SIP does not require any custom code to run on the IP-PBX. Each has their advantages for different scenarios.
Back to your problem, how come the From: header contains 9999 and not your SIP Profile ID? Or is that something you removed yourself when posting? Try calling their free test # +17606604690 to see if that works. -- Nabeel Jafferali X2 Networks Inc. -----Original Message----- From: Bruce N [mailto:[email protected]] Sent: June-11-10 9:30 PM To: Simon P. Ditner Cc: asterisk Mailing Subject: RE: [on-asterisk] Skype SIP Trunk settings for Asterisk - Anyone has it working? Thanks Simon. I do have 1 channel subscription and calls to New York and Toronto fails with this format: +14166356574 or 0014166356574. Would you have your trunk info for either registration or ip authentication that you can share? By the way, what was all the big fuss with Digium and chan_skype when Skype came up with this sip trunking so quickly? Doesn't this effectively kill the chan_skype? I don't see why anyone should bother buying their licenses anymore. Wondering if skype betrayed Digium or Digium just thought they should make some money on the side from the blind user. -Bruce > Date: Fri, 11 Jun 2010 21:00:27 -0400 > From: [email protected] > To: [email protected] > CC: [email protected] > Subject: Re: [on-asterisk] Skype SIP Trunk settings for Asterisk - Anyone has it working? > > Hi Bruce, > > You're right, it's just like other SIP providers. I've done fairly > extensive interop testing with them. Maybe you don't have any credits > assigned for outbound calling? > > On 11 June 2010 20:28, Bruce N <[email protected]> wrote: > > > > Hi Guys, > > > > Trying to setup Skype trunk but I am not successful. I do register but calls are met with Forbidden or 404 Not Found. I tried using the IP Authentication as well, and sip message comes back as this: > > > > --- (8 headers 0 lines) --- > > vps526*CLI> > > <--- SIP read from UDP:63.209.144.201:5060 ---> > > SIP/2.0 404 Not Found > > From: "9999" <sip:[email protected]>;tag=as244e0ffb > > To: <sip:[email protected]>;tag=ca90d13f-13c4-4c12d364-4bd2be89-50c0f 91 > > Call-ID: [email protected] > > CSeq: 102 INVITE > > Via: SIP/2.0/UDP 72.72.72.725060;branch=z9hG4bK12e4a53c;rport=5060 > > Content-Length: 0 > > > > > > This is the trunk detail I have: > > > > host=1.sip.skype.com > > > > type=peer > > > > > > > > I have 1 Channel subscription so the calls should go. > > > > > > > > If you have any FreePBX or Asterisk trunk details please share with me. Please note that this is chan_skype but rather simply Skype Business Control sip trunks which is I think like any other SIP provider. > > > > > > > > Thanks, > > > > Bruce > > > > > > > > > > > > _________________________________________________________________ > > Game on: Challenge friends to great games on Messenger > > http://go.microsoft.com/?linkid=9734387 > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > _________________________________________________________________ Turn down-time into play-time with Messenger games http://go.microsoft.com/?linkid=9734385 --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
