PSTN Caller ID Pattern and PSTN Access List Problem

[cliveburt]

After spending days trying to get Gateway dialing working to no avail, I've moved on to try and figure out which other parts of my VOIP "Master Plan" do actually work and which don't.

As part of my experimentation I tried a conguration that had Sipgate UK on Line 1 and Broadvoice on PSTN. Sipgate requires that you use STUN server settings. All there FAQ's talk about it, all their configuration examples include it. So I set it up using Voxilla Wizard....
Sipgate works fine, but whilst testing the Broadvoice side of things, on PSTN, I noticed that Broadvoice was registering every second! I checked the "Register expires" setting and every other setting that I thought might have anything to do with it...nothing seemed to effect it, so I thought I'd just carry on testing. I'd dial into the PSTN and try to enter a PIN to get VIOP access to BV. That didn't work! Hours of forum reading and trying solutions like DTMF settings and gain control got me no where.
Disabled Sipgate STUN settings...Broadvoice works fine!!!
I tried FWD's STUN server settings having read that one can use anyones STUN server, to see if I could get Sipgate and Broadvoice to live happily together...No Joy!
Broadvoice doesn't like STUN.

Move on to a different configuration...

This config still has Broadvoice on as PSTN (VOIP 2) provider.

I was trying to configure it so that all calls into PSTN line would ring through to the phone attached to Line 1 except if it's from my mobile phones number in which case it would give me a dial tone for VIOP access. I set "PSTN Ring Thru Line 1" to yes and entered my mobile number into "PSTN Caller ID Pattern" box. When I call in from my mobile it rings and rings on the Line 1 phone never asking me for a PIN or giving any indication that it believes my mobile Caller ID is Gods chosen one.
If i delete the phone number from the box, then it asks for a PIN when I call in or if the ex-wife calls in, or anyone else for that matter.
I even tried entering in the * wildcard into the box, figuring that that would have the same effect as having nothing in the box. No, all calls just carry on ringing through to Line 1 phone.
Does anyone have this working? If so, how?
Anybody have an surgestions on what might get it working?
I've searched this forum for answers and read the relevant parts of "SipuraSPAUserGuidev2.0.9pdf", which says it should work, all to no avail.

Thank you in advance for any help you might be.

Regards,

Clive

[mberlant]

First of all, STUN is a client function and not a server function. sipgate has no reason to require any particular NAT traversal method, since NAT traversal is the province of the client.

Sipura devices have two methods at their disposal for resolving NAT issues, and the one to use depends on the particular environment at hand. As you read the various threads here you will find several decent explanations of just how STUN works.

What you need to experiment with is STUN (on the SIP page - all STUN servers are perfectly interchangeable) and NAT Mapping Enable / NAT Keep Alive (on each Line page). Which scheme works depends upon your ISP and your SOHO router. It does not depend upon your VoIP service provider.

[cliveburt]

Thanks for that.
The first part of my post was an attempt to help out some poor soul who enters "Broadvoice regestering" into the search for the forum.

Did you have any thoughts on the CLI problem? All the STUN stuff was just an experiment but the CLI is more of a "must-have".

[mitussis]

This is a variation of a problem many of us seem to be struggling with.

We want to forward all calls except ones from a CLI list (which go to the PSTN->VOIP gateway). No one seems to have a way to do this easily without a firmware change. You might like to see the post about encouraging Sipura/Linksys to make the changes:

http://voxilla.com/forum-viewtopic-t-501.html

[cliveburt]

Mitussis,
You say in the other thread that you have "PSTN Caller ID Pattern" working!?!? How? As soon as I enter anything in that box none of the calls into PSTN get passed through to VOIP2.
What version of Hardware/Firmware do you have? Can you please give me an example of what you have entered into "PSTN Caller ID Pattern" box?

Thank you for taking the time

[cliveburt]

Has anyone been able to get "PSTN Caller ID Pattern" working? If so can you please what syntax you used?

[jcgalvezv]

Clive:

Instead of typing your Mobile caller id into PSTN Caller ID Patterns, insert it into PSTN Access List. PSTN Access list is a list of comma separated phone numbers. Numbers included in this list will bypass authentication process in SPA3000 and will immediately gain access to VoIP (according to default dial plan for PSTN-to-VoIP). Whenever you call from your Mobile, phone connected to Line port will ring for n seconds (those defined in PSTN Answer Delay. Look for it in PSTN Line tab, FXO Timer Values (sec) section) before SPA3000 answers the call for PSTN-to-VoIP gateway. If you want SPA300 to answer the call immediately (without ringing FXS Phone) set PSTN Answer Delay to zero, but keep in mind that if you are at home/office you wull not have a chance to answer the call.

Now, if you want to block any phone for PSTN-to-VoIP gateway, include this in any of the eight PSTN-To-VoIP Selective Call Forward Settings. If you want to block some number, lets say 123456 do this:

In PSTN User tab, PSTN-To-VoIP Selective Call Forward Settings section set Cfwd Sel1 Caller: 123456 but leave Cfwd Sel1 Dest: blank. 123456 will be blocked and SPA3000 will not even answer the call.

Juan C.

[cliveburt]

Thank you for taking the time to reply, Juan.
I understand and had already implemented your surgestions. However, it gets very very complicated trying block every other number, including "Anonymous", in the known universe, except my mobile number, using Selective call forwarding. Anonymous doesn't even work in the selective forwarding fields!
According to the manual all I should have to do is enter my mobile number in "PSTN Caller ID Pattern" and it would be the only number to be allowed through. That way I can set the answer timer to zero knowing that I'll instantly be allowed through and that everyone else will ring indefinately on Line 1.

So, back to the original question...

Has anyone been able to get "PSTN Caller ID Pattern" working? If so can you please what syntax you used?

[jcgalvezv]

Do this:

clear those fields (PSTN Caller ID Pattern and PSTN Access List) and call from your cell/mobile phone to your sipura 3000. While ringing see in SPA3000 Info page what number is calling. Then copy that number to PSTN Caller ID Pattern. I guess the number you have entered is not exactly the same SPA3000 is receiving.

Juan C.

[cliveburt]

Been there, done that
I've even used wildcards ie, *78112144*
I actually tried *7*.....
I tell you they might as well re-lable it to "PSTN-To-VOIP Gateway Enable:" = NO... it seems to do the same job!

Thank you for the surgestions. Keep them comming.