Thanks a lot. That sums it up. That is why I am getting the "SIP/2.0 403 Forbidden" for dialing other numbers. And it's good to know I can still dial toll-free numbers.
-Bruce > From: [email protected] > To: [email protected] > Date: Fri, 11 Jun 2010 22:07:51 -0400 > Subject: RE: [on-asterisk] Skype SIP Trunk settings for Asterisk - Anyone has > it working? > > The Skype for SIP Profile subscription gives you no calling minutes, you can > dial toll-free numbers and that's it. You can also associate a Skype > business account with the Skype for SIP Profile and receive calls from the > Skype network. > > > > To make outgoing calls to the PSTN, you need to add credit to your Skype for > SIP Profile and pay their per minute rates. And no, you cannot add Skype > subscriptions to a Skype for SIP Profile. > > > > -- > > Nabeel Jafferali > > X2 Networks Inc. > > > > From: Bruce N [mailto:[email protected]] > Sent: June-11-10 10:04 PM > To: [email protected]; asterisk Mailing > Subject: RE: [on-asterisk] Skype SIP Trunk settings for Asterisk - Anyone > has it working? > > > > 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 > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________ > > > > 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] > > > > _____ > > Your Photo on Bing.ca: You Could WIN on Canada Day! 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