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UK VOIP - Skype to SIP gateway

 
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Old 09-26-2009, 11:16 AM   #21
Default Re: Skype to SIP gateway


On Wed, 23 Sep 2009 18:25:59 +0100, alexd postulated:

>Roger Mills wrote:
>
>> But how many *English* - as opposed to *American* - speakers would want
>> her as the voice of *their* phones?

>
>How many English speakers are likely to give a ****?



Hirohito
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Old 09-26-2009, 11:18 AM   #22
Hirohito
 
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Default Re: Skype to SIP gateway
On Wed, 23 Sep 2009 , alexd ejaculated:

>How many English speakers are likely to give a ****?


I love good english ****

Hirohito <>

Nip


Hirohito
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Old 10-19-2009, 11:43 AM   #23
Graham.
 
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Default Re: Skype to SIP gateway


"Paul Cummins" <> wrote in message news:...
> In article <h950li$odf$>,
> (Graham.) wrote:
>
>> The preferred URL for my notes on the Skype to SIP gateway is now
>>
>> www.skypegateway.webs.com
>>
>> My son will be testing it during Freshers week starting tomorrow
>> morning.
>> Hopefully the cost of mobile phone calls will be one less thing for
>> him to worry about.

>
> Not working here for a similar setup...
>

OK, let's take it from the beginning. When you call the gateway,
is the call answered, answered and immediately dropped, or does it just ring.
Is there any activity in the SIPtoSIS window when you call?

--
Graham.

%Profound_observation%




Graham.
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Old 10-19-2009, 07:38 PM   #24
Graham.
 
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Default Re: Skype to SIP gateway


"Paul Cummins" <> wrote in message news:...
> In article <hbhfth$pku$>,
> (Graham.) wrote:
>
>> OK, let's take it from the beginning. When you call the gateway,
>> is the call answered, answered and immediately dropped, or does it
>> just ring.
>> Is there any activity in the SIPtoSIS window when you call?

>
> OK
>
> Call skype - skype identifies the call, and Sip-sis tries to route. I
> hear nothing on the skype phone, and 25 seconds later, the call fails.
>
> Without pbxes online, the call simply fails, so I guess it's in there
> somewhere...


Situation seems retrievable
Make sure the "trunk name" you have given to the orbtalk trunk is identical
to the "trunk" in the inbound route.

Also in the inbound route, check the "caller ID number" is blank,
The route name will have a trailling slash eg. orbtalk/
If you accedently create a route that appears just as a slash by itself, delete it.

The destination is set for regular hours to "callthtu PIN"
and "Force regular hours" is selected lower down.

Try dialling directly into pbxes by dialling the Orbtalk number


--
Graham.

%Profound_observation%




Graham.
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