[cisco-voip] Why is my CUBE sending my dial-peers instead of login information to Skype Connect?

Stephen Welsh stephen.welsh at unifiedfx.com
Thu Mar 3 08:28:01 EST 2011


Hi Nick,

Set your 'Calling Party Transform Mask' on the outbound Route Pattern in UCM to your SIP Connect ID (i.e. 99.......)

That should set the From Header in the outbound request so Skype will accept your call

You can use 'debug ccsip message' on the CUBE gateway to check the FROM header is being set correctly.

Thanks

Stephen

On 3 Mar 2011, at 04:08, Nick Matthews wrote:

I would put a no sip-register on your pots dial peers to make it clean.  All those guys are going to try and register by default.

If you have ephone-dn's you can put no-reg on the number as well, same deal.

You may want to create a dummy pots line or ephone-dn with the correct From: DID to get it to send the register message.  Then you do some translation/forwarding to get it where you want it.

-nick

On Wed, Mar 2, 2011 at 9:30 PM, Robert Kulagowski <rkulagow at gmail.com<mailto:rkulagow at gmail.com>> wrote:
On Tue, Mar 1, 2011 at 3:33 PM, Stephen Welsh
<stephen.welsh at unifiedfx.com<mailto:stephen.welsh at unifiedfx.com>> wrote:
> I've configured Skype Connect using a CUBE config (SIP2SIP) directly on the internet, I did it for the same reasons are yourself, however I was disappointed at the cost (to actually place calls) and the fact you cannot call to a Skype User ID from the IP-PBX (you can do the reverse by calling your Skype Connect ID number from any Skype Client). I was hoping to do some Single Number Reach with my Skype ID :(
>
> Below are the relevant parts from my working config, however you need to add your own security statements (I'm not responsible for your phone bill :), you will most likely be exposing port 5060/61 to the Wide Wicked Web, and I've seen a LOT of registration attempts hit my router....
>

So, after a "wr erase" and a reboot, then starting from scratch, I've
at least got the REGISTER part sending my Skype Connect username
rather than my dial peers.

The next part to figure out is if they're rejecting my "from"; it
looks like when I send an INVITE to them I send my internal 6-digit
extension and I end up getting a reject.  Is there something else that
I'm missing from my config?   Are they expecting the FROM to be my
Skype ID, with the TO being the e.164destination at sip.skype.com<mailto:e.164destination at sip.skype.com>?

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