Thanks for that Nabeel.

 

9999 is the x-lite extension that I am placing the call through and it was 
showing up since I didn't have Caller ID setup as the Skype username.

 

The +17606604690 works fine with the registration mode (I didn't try IP 
authentication mode yet).

 

I guess that brings me to the question if Skype allows dialout with their 
Subscription plan or not?

I know they don't allow that with Asterisk when it comes to chan_skype. I 
thought they allow that with their sip trunking process. They have the $6.95 
channel/month. My calls to other destinations are met by 403 Forbidden.

 

Can you guys confirm if that works for you or if only their per minute call 
works?

 

Thanks,

Bruce
 
> From: [email protected]
> To: [email protected]
> Date: Fri, 11 Jun 2010 21:54:34 -0400
> Subject: RE: [on-asterisk] Skype SIP Trunk settings for Asterisk - Anyone has 
> it working?
> 
> 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
> > >
> > >
> > >
> > >
> > >
> > > _________________________________________________________________
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