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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

How to make true IP-to-IP call with RT31P2 ?



 
 
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  #1  
Old December 10th 05, 08:07 PM posted to uk.telecom.voip
peter
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Posts: 49
Default How to make true IP-to-IP call with RT31P2 ?

How do you make a true IP-call from an RT31P2 to an RT31P2 ? (the
innards are more or less like as Sipura 2100) ?

If both are registered with a provider, is there any way to by-pass the
provider without too much fiddling with the set-up, so as to make a
direct IP-adressort - IP-adressort call ? If so, what settings do
you neet to set ? Are "Make call wihout reg" and "Ans call without
reg" sufficient ?

After some experimentation, I was unable to bypass my provider in
anyway, i.e. anything I did seemed to be routed thrugh my VoiP-provider
(because, on all attempts, the recipient received my phone number as
Caller-ID, and that info is only stored with my provider).

Anyone with any experience on this subject, or some good advice ?

Thanks

peter

  #2  
Old December 11th 05, 02:09 AM posted to uk.telecom.voip
Martin²
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Posts: 848
Default How to make true IP-to-IP call with RT31P2 ?

You can dial directly any IP number, just start and end with hash and
replace full stoops with asterisks e.g. #123*45*67*89#
If there is a SIP device at that IP address it should ring.

Mind you if your RT31P2 is locked to Vonage or some other provider, then it
may not work.
Regards,
Martin


  #3  
Old December 11th 05, 07:39 PM posted to uk.telecom.voip
peter
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Posts: 49
Default How to make true IP-to-IP call with RT31P2 ?

The issue is not how to execute the dial procedure, but how to avoid my
own provider :-) All my attempts were in fact registered by my
provider, and the whole point of the exercise was to bypass him

  #4  
Old December 11th 05, 07:47 PM posted to uk.telecom.voip
peter
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Posts: 49
Default How to make true IP-to-IP call with RT31P2 ?



You would need to have 'yes' for these settings.

I did .

'Enable IP Dialing:'
would be 'yes' if you intend to dial [email protected]_address using the
handset. That's assuming these Sipura type settings are available on
the RT31P2.

It is and I did. Although, Í had the IP stored in the Speed dial, and
tried also in the dial plan.


What was the nature of the experimentation? I would have unregistered
the line as part of a test.

Initally i tried while registered. Then I tried the speed dial wile
unregistered (but without erasing the provider info).

Also, my experience is that the caller ID
is the User_ID:

Nope - not with my provider. The user-IDs are composed of letters and
numbers, because my provider allows to sperate lines on the same phone
number (with two different user -ID) - hence, the only place where the
link exists between my user-ID/IP-address and phone number is with the
provider.
Vonage ?

No not Vonage - some small outfit abroad

Also, with regard to unregistering : The whole point was to find out
how to be able to use my normal provider AND, occasionally, dial an
IP-to-IP call, without have to refiddle all the settings each time
Perhaps it is just not doable on my Rt31P2 ?

Anyone managed such a feat ? In which case, which hardaware did you use
?

Anyway - thanks for your comments Brian.

  #5  
Old December 14th 05, 08:45 PM posted to uk.telecom.voip
alexd
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Posts: 1,765
Default How to make true IP-to-IP call with RT31P2 ?

peter wrote:

The issue is not how to execute the dial procedure, but how to avoid my
own provider :-)


All my attempts were in fact registered by my
provider, and the whole point of the exercise was to bypass him


SIP clients are designed to 'reinvite' calls - i.e. bypass the SIP proxy
[your provider] and connect directly to each other once codecs have been
negotiated. However, your provider can disable this functionality if they
decide to, for example to capture DTMF digits, or [rarely] to transcode
data between clients that don't have compatible audio codecs. So in theory
[and in all probability; why would your wafer-thin-margin telephony
provider want to trombone data in and out of their SIP server, or waste CPU
cycles on transcoding audio?] your RTP stream is going from your phone
directly to your callee's phone without going through your provider. Your
provider is likely only doing the call setup.

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