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AFAICT the entire 'Call extension' feature is somewhat lacking. First of all, you cannot enter a Hunt Group. Then, you can only enter a SINGLE extension as a replacement, so you cannot have the call being directed to several extensions. This is particularly aggravating since I'm sure I read somewhere in the docs about being able to enter several extensions (just separated by a space) as in other ares of the PBX config, but it doesn't work. Since this is already due for an overhaul because the Test system is unable to test any 'Call extension' lines in a dial plan, at the same time, can we have them make the very sensible inclusion of being able to send the call the multiple extensions and/or Hunt Groups?

er, doesn't. If I include a line in my dial plan that has the above selected, even though a successful match is made, the result in the test area just shows the following error:- "No pattern found for the dialed number" If I switch to sending it to a trunk, it all works, but if I want it to Call Extension, it fails. I followed the docs and just used the intended recipient extension number as the replacement. I also tried creating a full sip URL, but neither works. I must be missing something. Can anyone explain what?

No idea as it doesn't do what I actually need. I have several problems with this. The first is that if you use square brackets and 'x's in the same pattern, it only works when the square bracket expression is at the end, but NOT the other way around. This would appear to be contrary to any sensible pattern matching algorithm. Here's what I want to do: match any 6 digit number that begins with 2,3,4,5,6,7 or 8. In the end I used "^[2-8][0-9]{5}@" which works as I have forced it to used ERE rather than it being just a simple expression and I figured for myself that the SIP stuff IS added prior to attempting matching. None of this is rocket science, but when coming to it anew, decent documentation is essential. There are so many omissions and errors in all the snomONE docs I've seen it's not funny. Anyway, now I have a better understanding of how snomONE does its pattern matching and I ought to be able to achieve what I need to do with dial plans. However, here's an idea. Cascading Dial Plans. IOW an extension can specify several dial plans which run in the specified order. In this way, you could have a standard dial plan that was used by several extensions for some basic matching, then if that 'fails' it rolls over to the next dial plan which is maybe specific to that particular extension. This makes it easier to plan and maintain as any modifications to the common dial plan are instantly implemented for ALL extensions that use it. Currently, if you have a lot of similar matching in many dial plans used by many extensions, if you need to modify any of the basic initial matching, having established what needs to be done, you then need to update each and every dial plan that uses those same patterns. I like this more object oriented approach with maximum re-use of 'code'.

Isn't there some way to force the PBX to use ERE or not? If we don't have this control, it makes configuration that much more complex since we have to always guess at how the PBX will actually interpret the pattern. Hardly ideal. You say the input format is user@domain. Can you explain this. As far as I am concerned, the user enters a number on the telephone keypad and I want to control how that number is maybe modified and then used. I assumed any SIP stuff was added later. Are you saying that after the number is entered, the SIP stuff is added and THEN it is matched to the dial plan? Is there somewhere a full and complete and accurate explanation of dial plan operation that I can access and read? What I have found so far is woefully inadequate at explaining it all, meaning I have to keep asking. I can deal with Regular Expressions if I have all the details. Right now I feel I'm blundering around in the dark with no documentation to enable me to see the light.

I am testing a dial plan and becoming increasingly unable to fathom the matching logic. As an example:- xx[2345678] matches 358 as it should, but not 359. Again, correct. But [2345678]xx does NOT match 358 which it obviously should. Can someone explain what's going on here? Also, where is the full explanation of Regular Expression Matching? Since this can vary between implementations, it would be good to have an exact description of how it works in snomONE. I'd like to know what I can and cannot expect to work. E.g. [^8] should match anything except 8 in 'most' regex, but doesn't work in snomONE. I can work with such differences, but would appreciate some documentation to explain this, rather than make me simply test everything to see if it works.

You assume I know what the 'other' method is and that's the problem, I don't. Anyone can dial into any VM and access messages etc IF THEY ENTER THE CORRECT PIN. This is not what I would call a 'Shared' or 'Group' mailbox which should allow access from any of the extensions configured to use that shared mailbox as if it was their own, i.e. without having to enter the PIN. The manual appears to explain how to do this, just enter the extensions that can share the mailbox into the 'Allow Access for Extensions:' field. But having done that, it seems to make no difference and dialling in from those extensions still results in a need to enter the PIN. Now it's possible that Snom's idea of a shared mailbox is different to my own and that they consider having to enter the PIN to access it is acceptable, but that seems unlikely since you get that facility WITHOUT entering the other extensions into the 'Allow Access for Extensions:' field. Configuring the other extensions like that obviously SHOULD change how it works in some way, but as I said, the manual give NO indication of what that change is. If it is to allow easy access by the other extensions, i.e. no PIN required, then there is a problem because it doesn't work. If the aim is something else entirely, then I would like to know because the manual doesn't give any idea of what that is. So, first of all, should a 'Group' mailbox allow access without PIN to those extensions specifically listed in 'Allow Access for Extensions:' ? If so, why doesn't it work?

I beg to differ. I refer you to page 301 in the PDF manual, i.e. Chapter 24: Voicemail, in which is mentioned Group Mail, i.e. a shared Mailbox that can be accessed by multiple extensions as I described above. But in the manual it simply states that: "The messages that are left in the group mailbox can be picked up from either the user’s extension or the group mailbox extension." and no mention is made of having to enter a PIN. The latter being the process by which anyone can pick up voicemail from any extension if they know the PIN. This is not how I would understand a shared Mailbox and it also somewhat contradicts the description in the manual. Typically the explanation in the manual however is very brief, to the point of missing out any real information about how it really is supposed to work. It is this information I seek and as usual some guidance from Snom wouldn't go amiss.

I'm trying to set up a shared mailbox, but can't even work out how it is SUPPOSED to work. Let's say there are 3 extensions (40, 50 and 60). 50 and 60 have their own mailbox which sends out voicemail as email attachments and deletes the message. This works fine. The other extension (40) needs to be a shared mailbox so that 50 and 60 can also retrieve the messages which are NOT emailed. I have added "50 60" in the 'Allow Access for Extensions:' setting in 40's Mailbox section, but what is that supposed to do? If I dial 840 from either 50 or 60, I seem to get exactly the same as before and asked to enter a Mailbox access code or something like that. Whatever it is, I cannot find any setting with the same name. But in any case, I just want 50 and 60 to be able to immediately access 40's Mailbox without having to enter any special code - whatever it's called. Can anyone clarify the shared Mailbox function?

No contradiction there at all. I'm just preaching freedom from control. To view that as some sort of control in itself is one of the great philosophical fallacies. As regards other customers requesting the validation check, it saddens me that they have requested a feature that actually prevents the use of valid email addresses. It's just the way of the world these days. CONTROL everything - not because it's necessary, but just because you can. The ability to modify the validation function is however a truly great feature and I'm delighted to find it's included. Where would I find this mentioned in the manual?

Ah thanks. I'll try that.

When an email address is entered, it is checked and rejected if the PBX decides it is not valid, which means it is impossible to enter addresses which do not contain the @ and I think maybe it likes to see at least one period. Anyway, the problem is, in my local network, there is NO domain part required, just the username. In fact, it is kinda tricky to make it work with any domain part. For local email, it is just sent to the username. But, snomONE WILL NOT LET ME add this as an email address. First of all, I would point out that that the username on its own is an entirely valid email address and works PERFECTLY once I have edited the relevant XML file, but this is a PIA (not least because of the total lack of readable formatting in pbxnsip's XML files - a daft omission). Secondly, I firmly believe it is not appropriate to attempt to FORCE users into adopting the practices perceived as 'correct' by the developers, even when correct, but especially not when those beliefs are themselves invalid or inaccurate. Sorry, but this is a case of TOO MUCH CONTROL. So, please, could we not be allowed to enter whatever we want as an email address? If it's wrong, it won't work and we correct it. What's so bad about that? Why do we need these idiot checks by the software developers - as I said, particularly when they GET IT WRONG. Perhaps some support here would convince snom of the error of their ways so they remove such checking and let us enter any email addresses we want.

It has been stated snomONE free will now allow 5 non-snom registrations. Unfortunately, when you try to register a third non-snom device, it is rejected just as before with the same error message in the log file. I don't know if this is just Mac specific, but I couldn't find anywhere better to post this problem. Anyone else finding this? Is this a deliberate ploy or just another er, mistake? Snom?

ASAP please.

Oh yes. If I select to view 'All Topics I've started or posted to', there is at least one Topic to which I've posted several times (specifically the Service Flags in Dial Plans to which pbxnsip just replied) that is NOT listed. If I select to view 'All my posts', then my posts to that Topic (and maybe others) are not displayed. If I click the 'View New Content' link at the top, then that Topic IS the only one shown, so it does seem to know I have posted to it, so why is this Topic not displayed in the 'My Content' list?

Well I agree we certainly want to avoid instability, but that applies to any development and this would be such a great feature. Your previous posts on this subject led me to believe that it was due soon. I'm not sure what you are now suggesting. When might we actually see this feature?

Well, certainly not Ghost Busters. In fact I'd quite like to contact a forum admin because something is not working correctly (some of my posts are not appearing in 'My Content'), but to whom are we supposed to report such errors? I can't find any links anywhere to do this. I tried one of the Snom support profiles (pbxnsip) and although invited to send a message, it just results in an error saying they can't receive any more messages. I'll avoid the obvious jibe about Snom not even able to manage their own support forum (oops, I just made it:-) and just ask that someone explain how to report such problems.

We've just had a new update, but still no Service Flag option in Dial plans. This would be such a great feature and you say it has been implemented, so please let us have it. Why the delay? Please explain what is the situation regarding this feature.

I think you are misunderstanding the SPA3102. It is NOT a simple piece of kit to get your head around. First of all, NOTHING registers. So if you have either the SPA or snomONE trying to register to the other, you've got it wrong. For the Gateway Trunk in snomONE Name: whatever you want Type: SIP Gateway Direction: Inbound and outbound Trunk Destination: Generic SIP Server State: Enabled Display Name: anything you want to appear on outgoing calls Set Account: and Username: to any number you want and use same number in SPA config below instead of ?????????? Domain: FQDN of your SPA or its IP address. Password: whatever you want Proxy Address: Domain:port (i.e. same info as in domain, then the port number you use in the SPA setup, probably 5061. Don't forget colon separator) Explicitly list addresses for inbound traffic: Domain (i.e. same as in Domain, but no port. This helps ensure incoming PSTN calls use this Gateway) All SPA config is done on the PSTN Line tab. The Line 1 tab is purely for setting up a connected analog phone to use VOIP. If you only want the SPA to act as the PSTN Gateway for the PBX, you can leave Line 1 disabled SIP Settings - SIP Port: something NOT 5060. Default is 5061 and should work for you. Proxy and Registration - Proxy: FQDN or IP address of your PBX - Outbound Proxy: BLANK - Use Outbound Proxy: no - Register: no - Make Call Without Reg: yes - Ans Call Without Reg: yes Subscriber Information (all blank except:-) - Password: use the password you set in snomONE - User ID: ?????????? - Use Auth ID: no VoIP-To-PSTN Gateway Setup - VoIP-To-PSTN Gateway Enable: yes Dial Plans (use any, e.g. 2 - you MUST include the enclosing brackets below) - Dial Plan 2: (<:??????????>S0) PSTN-To-VoIP Gateway Setup - PSTN-To-VoIP Gateway Enable: yes - PSTN Ring Thru Line 1: no (unless you have a regular phone connected to the SPA and want it to ring) - PSTN Caller Default DP: set to whatever Dial Plan you set up above, e.g. 2 - PSTN CID For VoIP CID: yes (this will pass incoming caller-ID on to the PBX. BUT will cause failure with UK VOIP providers. In which case, choose no) International Control - SPA To PSTN Gain: 18 - PSTN To SPA Gain: 12 Be aware that the SPA is sensitive to its 'gain' settings. It's easy to make call volume too low, or introduce bad echo if set too high. I've provided you with my current Gain settings, but you might need to play with the settings to optimise call quality. All other International settings (and e.g. in the Regional tab) can be adjusted to suit your local conventions. It's likely you can find appropriate settings from the 'Net that will ensure call ringing etc sounds correct for you. I think that's it. This is the correct way to set up the SPA and it does work. If your's doesn't, then either you've not copied my instructions correctly, or I've missed something out (which is quite possible, but I hope not:-) I hope this helps.

I understand. It is only for about 10 actual clients so looks like a Hunt Group is the way to go. I thought it would be, but thanks for the confirmation.

What is the best way to make an Internal call to ALL other extensions? I don't think there is any inbuilt function to do this. Paging is not what I'm referring to here as that is a one way process. I want to call all the other internal extensions so that someone can pick one up and answer the call as per a normal call. I was thinking of setting up a HuntGroup that included all the extensions that I want, just for this purpose and while I believe it will do what I want, does anyone else have a better idea of how best to do this?

I have to say that changing the ports is mentioned somewhere in the instructions, but it is not made sufficiently clear or even easy to do since you have to delve into pbx.xml which is inconveniently totally unformatted. In fact, IMO they should supply it with a default setting that will work and then advise users to change them back to std. if that is what they need. At least with the PBX running this can be easily done via the web interface. Still, glad I could help.

Have you set HTTP and LDAP away from the std. ports? If you haven't done that it WILL crash on startup. Just change them to ports NOT being used by OSX Server.

Do you mean matches just Username, or would it match an Account name? Or any other field/parameter?

That's my point, I cannot see how it could possibly match. Account, Domain, Username, Password, Proxy match NOTHING in the incoming call data. Whereas in the PSTN Gateway Trunk I have explicitly set the address from which to accept calls which according to your rules above would produce the highest ranking match. So why send it to a Trunk that apparently doesn't match. Ah. It is possible the test trunk did not have a Proxy set at the time it was assigned those calls instead of the PSTN Gateway. I recently corrected this in one Trunk and it could have been that one. It would also explain why current test calls seem to be handled correctly. However, what you're saying is that an empty Proxy setting will match anything, with a higher ranking than any of the rules you state above, even higher than a Trunk with a matching proxy and an explicitly stated address?

Outbound is controlled by the Dial Plan so no problem. Inbound is the problem and only for calls from the Linksys SPA3102. Calls from the VOIP provider seem to always be assigned the correct Trunk, no doubt due to the use of the 'Line' parameter when registering.