DaveD

As I clearly explained, all I have to do to get the service to start properly is change the .EXE file--not the log file. This means that the log file must have the required '$'--and it does: DaySys-log_$.txt When I follow the WIKI article to create a log file, the response is [in a pop-up window]: The procedure entry point K32GetProcessMemoryInfo could not be located in the dynamic link library KERNEL32.dll This seems to indicate a fundamental failure in the compiled image. And the executable failed to start; as you'd expect, the service will not even start manually. When I change back to the 5.1 .EXE, the program starts instantly, and appears to create an ordinary log file. This action didn't start the service automatically, but I started it manually with no problem. This SBS server is the domain controller; it hosts only one domain. No trunks are global. Dave

No, neither reply indicates you have read my post. First, this is SBS2003--not WindowsXP, or any other desktop Windows OS. I can't say if VS2014 compile is compatible with this--can you? And this is (as I said) about the service failing to start-- after an install when appears to complete normally. So manual upgrade should have no effect on this. Again this is about the service failing to start--not about the Web interface either. Finally, I mentioned the very long-standing issue with multiple Vitelity trunks failing to work. There was no mention of this in your replies. Can you please respond to my post? thanks, Dave

PBX is installed on SBS2003 [which is 32bit] Installation of 5.2.1 or 5.2.2 proceeds normally, but fails to startup. I find that the service is stopped, and it will not manually start (it 'times out'). When I replace the .EXE with the original 5.1.3 file, the PBX will then start and run. I suspect (but I have no way to prove) that the URL update strategy only loads a 64bit image. Here are the file sizes (as shown by Windows Explorer): >5.2.2 5,498,880 >5.2.1 5,454,336 >5.1.3 7,493,680 Note how much smaller the 5.2.n files are. I can't find any way to directly install a 64bit image, because it can't be downloaded; only a fresh install will provide that image, and would wipe out this one. [bTW: I have never been able to run two Vitelity DIDs at the same time. I was able to kludge the PBX to run one DID by splitting inbound and output into two different trunks. But whenever I add a second DID [either as a single or a split trunk], the other trunk fails to handle inbound or outbound calls-- even though the GUI shows no registration issues. I've more or less given up; I've spent a huge amount of time--and provisioned two trunks for Vodia to test.] What do I do? Dave

Thanks for an incredibly quick reply! My ITSP specifically recommends not to set an outbound proxy; that's why I didn't set one. They explained when I asked that it's about their servers not being load-balanced for in/out calls. Because I do have the ITSP SIP server specified, I thought that would be the only inbound route, but I'm apparently wrong. Where do I 'specify the IP address where the trunk expects traffic from'? Dave

I'm receiving bizarre calls I believe are an attempt to pass toll calls through my server. Several inbound calls on my SIP trunk are show as from '100 (100)' in the call long. And the 'to' field shows one of these: 00972597841671 (00972597841671) 0972597841671 (0972597841671) 9011441904898504 (9011441904898504) 011441904898504 (+441904898504) From this format, it appears someone is trying various formats to dial either Israel or the U.K. There is no extension 100 registered, and because these calls apparently ring local extensions, no external call has actually completed. But because the call log shows an invalid 'from' and a 'to' that may be a valid international number, it appears there is some external access to the server. This really concerns me; what's going on here? Dave Here is a segment of the SIP logfile: SIP/2.0 200 OK Via: SIP/2.0/UDP <server local IP address>:5060;branch=z9hG4bK-22c1c73a8c11f54f47aaffcf117679bc;rport=5060;received=192.168.10.15 From: "100" <sip:100@pbx.company.com;user=phone>;tag=24006 To: "00972597841671" <sip:00972597841671@<public IP address>;user=phone>;tag=635631766 Call-ID: 7d6ccdf3@pbx CSeq: 30725 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Server: Aastra 9480iCT/3.2.2.3077 Supported: path Content-Length: 0

Did you folks find any use for the document I created and provided above? I hoped you might find a place for it in the WIKI. Dave

I sent you the complete log messages, and also the dial plan, fully annotated. As tech support requested, I also provisioned a Vitelity DID (SIP trunk) for your exclusive testing use. You did evaluate the SIP traffic log briefly, and said there are important aspects missing. You said a week ago that you would do further testing. This PBX is still unable to use multiple trunks; have you made any progress? thanks, Dave

Thanks for taking the time to reply! What you say is that only Snom brand phones have any support documents. It's not possible to configure any other brand phones. Some other brands have PnP, but no document explains what features or buttons on those phones are enabled or work. It's not clear how to setup contacts, address books or screens on any but Snom brand phones either. There's no way to create a customized config file, because button and feature mapping isn't published. For non-Snom brand phones, this is the only info for all PnP customizations; there is nothing more: Currently, plug and play is supported for some non-snom phones under snomONE from v5x on. Please tell me if I'm wrong about this. thanks, Dave

Actually my inquiry was very specific--but it does cover several areas--all directly related to SnomOne setup for phones. These answers are very specific to SnomOne--they're not available on the Internet. I haven't found anything of value in the WIKI with enough detail to be any help. Would you please look at my post again and reply? Dave

Right: the manual dates from 2010. In the mean time, consider dropping this example into the WIKI; unfortunately, plain-text paste doesn't preserve tabs: <<<<< SnomONE simple dialplan for NANPA [North American] dialing, using 10-digit format, with prefix to select a trunk [DID] Assume you use a main trunk [Trunk 'A'] for most outgoing calls, and two other trunks ['B' and 'C'] that you use as alternates. You also want to verify that the dialed string is exactly exactly 7 or 10 digits as required by NANPA format; any other length string is invalid and is rejected. International calls are an exception; they have variable length. If you dial a 7-digit call, your local three-digit area code must be added. In this example, we use area code 617. In this example, you can select one of the two alternate trunks using a prefix--either '9' or '8. SEQUENCE TRUNK DIAL PATTERN REPLACEMENT NOTE 100 N/A 211|311|411|911 not allowed 200 N/A 900xxxxxxx|976xxxxxxx not allowed 300 A 011* international 400 B 9([0-9]{7})@.* sip:617\1@\r;user=phone 500 C 8([0-9]{7})@.* sip:617\1@\r;user=phone 600 B 9([0-9]{10})@.* sip:\1@\r;user=phone 700 C 8([0-9]{10})@.* sip:\1@\r;user=phone 800 A xxxxxxx 617* NANPA 7D 900 A xxxxxxxxxx NANPA 10D NOTES > the sequence is crucial; for example, it's necessary to check for the '8' or '9' trunk prefix before strings without a prefix; otherwise it might be mistaken as the first digit of an actual phone number > 100: all the special-service [3-digit] numbers are disallowed > 200: the fee-based 900 and 976 exchanges are disallowed > 300: International calls start with '011'; here, only Trunk 'A' is used; the replacement string is simply 011 followed by any number of digits > 400: the pattern 9([0-9]{7})@.* selects outbound trunk 'B' and accepts exactly 7 digits (after the '9'); the replacement string is: sip:<outbound domain>617nnnnnnn;user=phone [the 'n's are 7 digits as dialed] > 700: the pattern 8([0-9]{10}@.* selects outbound trunk 'C' and accepts exactly 10 digits (after the '8'); the replacement string is: sip:<outbound domain>nnnnnnnnnn;user=phone [the 'n's are 10 digits as dialed] > calls that don't fit any dial pattern are rejected DaveD -- Sep13 >>>>>

Thanks for that! I've sent you a PM with log and dialplan attached. Note that the visual log in the domain log screen shows very strangely; some of the text is full size, but most of the SIP details are in very small type, and unreadable. This happens both in Chrome and Firefox. If I copy and paste to Notepad [plain text], all text is full-size. (The log file I sent is in plain text and doesn't show this issue.) luck, Dave PS: trunk I provided is currently set to 'ATA' type routing.

Thanks for a good discussion! And of course I respect the need for revenue from development time expended. We didn't ever address the UCClient for Windows problem; it doesn't work--at least not here; it won't even register. And you say you won't support it any more. Yes: I read a lot about this industry; of course there is change, and vendors are jockeying for the best position. Even so, we are still stuck with messy, incompatible legacy SIP 'solutions'; even the basics haven't been fully resolved. And whether WebRTC is an industry tie-breaker is unknown. Frankly, a user doesn't care how the code is delivered; if (for example) you can deliver a frameless Web applet that's locked into a small window [visually like the UCClient], that would be just fine. That would leverage WebRTC or any other browser-based code you prefer. But yes: this Web browser is clunky--it's close to unusable-- simply because it takes up far too much screen space and is functionally incomplete. Theres a clear need for a Windows- based feature-rich app; the market only wants a good answer-- they don't care how it's done. Dave PS: For an Android smartphone, I use CSipSimple; it's free and works well. [Yes: it's open-source.] And CounterPoint used to have a free limited version of Bria until they got greedy. For PBX users, the maker beginning with a '3' has a free full-featured Windows softphone with roaming integration to the PBX, presence and a synchronized phone book. All of these are compact, visually appealing and useful.

No, these trunks are not only for outbound traffic, as I already pointed out. I tried setting up separate accounts for each trunk--one outbound, one inbound. That didn't resolve this issue. If I'm 'confusing' the ISP with registrations, why didn't the Epygi appliance I used for years have the same issue? These are the same trunks--same accounts. I've also used other software PBX without this problem. This is not an issue about CID; I never even mentioned that. I do have questions about CID--for another thread. As you say, probably not a Windows or license issue. But I'm left with a system that doesn't work, while my old junky appliance does (although it has other issues). Isn't it likely the log files will tell you what's wrong? Dave

This was a new installation--not an upgrade. BUT: I recall that I had this same killer issue when I installed earlier versions: 4 and 5.0 (on different hardware). That's why I abandoned the product back then. As I already explained in my first post, this is a single carrier, on the same (authentication) domain but using different accounts. Again: there's not one account; if I PM you the logon info and you logon, that will lock out the accounts here. That's not a great idea. Isn't there some log info that will tell you what's going on? This is--generally speaking--a very ordinary 5.1.0 installation, except perhaps that it is on Windows SBS server 2003 (32bit), using port 81 as access. [i can't get SSL port 444 to work.] And again, you're right: I've had this exact problem before with earlier builds. It's likely a trunk compatibility issue, because others have reported similar problems with Vitelity trunks. But Vitelity is a major ISP... Dave PS: I find that if I disable all but two trunks, only one of the two is active; the other responds with fast busy, as if the trunk were out of service.

Um--got it! So doesn't that indicate a clear need for an app that fills this essential purpose? Dave

Thanks much for this! You provided some notes that try to make sense of the example. And to some degree, so did that 'oldie' you quoted. But not all symbols are defined For my system, '\d' did not work--it was passed as a literal--I had to use '\r'. I thought I'd found all the on-line dialplan sections, but because there are four separate ones--each slightly different--I missed this old one. And this old one is the only one that attempts at a definition. Even here, there is no explicit definition of some of the symbols or some of the syntax. For example, those '\d' and '\r' switches are not defined. I also added a 10D version of this as: 8([0-9]{10})@.* >> sip:617\1@\r;user=phone For multiple trunks, I use selector prefix of 9, 8 and so on, as you'd expect. Note that these four examples: > prefix 9 with 7D allowed > prefix 9 with 10D allowed > prefix 8 with 7D allowed > prefix 8 with 10D allowed are all the common examples of trunk selector prefix needed in a NANPA environment, and they don't appear in complete form anywhere. Why not publish them in the WIKI? Not to be a total nag, but doesn't this show the need for a single well-compiled document that shows all the syntax with examples fully explained? I'm a veteran tech writer, so perhaps I emphasize this. But good documentation IMHO is part of the core job--not an afterthought. Dave

Thanks! Please explain what the pattern does and the meaning of these symbols, so I can understand how to use the example in various actual installations. While 'regexbuddy' may be a good tool, it costs $40. Vodia really needs to document the syntax it uses, so ordinary intelligent people can use it. Dave

Thanks for your answer! Bria is now $50 a pop--that's a lot. Will it support the features I mentioned? I had already logged in as a user, with Chrome. But (as I said) that takes up much of the screen; it's not a compact elegant applet in the corner like UCClient (which I can't get to register.) It's not suitable for a full-time desktop app. And (as you said), it's not fully implemented. Dave

Thanks for your reply! I've looked at those samples for a long time; they're what I referred to in my post. Of course there's no programming language, but there is syntax that isn't well explained. I'll wait for that solution you offered to send me for a particular issue. But we need a document. Dave

I've been doing VoIP telephony for a while, and I thought I understood dial plans somewhat. But I find the sections here on SnomONE dial plans close to indecipherable. There's no complete list of symbology and syntax; there are only hints and incomplete examples. Yes: there are some basic plans, which are obvious. But the more advanced plans talk about the usage and show formats, but don't explain what the resulting output is, so you can't infer much. Examples are both incomplete and not well explained. I have a need for a very common extension and NANPA handling dial plan syntax, but I can't figure it out. And the SnomONE syntax isn't exactly like any other; for example it's not the same as Asterisk. (That's fine with me!) But without a syntax glossary, I'm just guessing. Dave

There's a clear need for a simple Windows extension applet that takes very little desktop space, and provides both making and taking calls, as well as IMs, voicemail and shows presence for all extensions. SnomONE has apparently deprecated the UCClient that provided most or all of these features. And although the client Web login provides several of these features, it is bulky and not suited for use as a full-time desk applet. What should we use as a desktop phone instead? Can a softphone such as CounterPath Bria [at $50 a pop] do this? Is there a multiplatform / Android app that will work? Dave

I use a variety of phones as extensions, and of course I want to be able to use buttons and lamps as needed. I can't find any complete document that shows how to do this. For example: > in general, what phone features / settings does SnomONE support? How do you select and map specific features? > how do you make a custom config file for a specific phone? there are a few SNOM phone files, but no notes about how to create your own file for any brand phone > what is the file format for PnP? I think it's html, but what is the internal format? > how do you map a button to a function / feature? for example, how do you determine the button's internal number? What set of features is available? > how do you config shared appearances; for example, - a busy-lamp field [bLF]? - shared line appearances? > how do you use a smart screen such as several phones from Grandstream, Panasonic, Yaelink and others have? I assume there's an XML mapping scheme, but where? How can you pass-through Internet/Web applications to the screen? > how do you display a scrollable / searchable contact list, with edit features at the phone? I realize there are bits, pieces and hints in various places, but it takes forever to glean this stuff. There should be a document. Dave

I've configured multiple SIP trunks: > each trunk both inbound/outbound > different DIDs > different [unique] login credentials > same carrier [Vitelity] > same [Vitelity] domain; same proxy [outbound.vitelity.net] Each trunk authenticates OK and accepts inbound traffic OK BUT: > if all trunks are enabled, apparently no trunk accepts outbound calls [this is not a dialplan issue] > if only one trunk is enabled, outbound calls are OK I've used these identical trunks, configured in the identical way on a PBX appliance box for years, and they worked outbound OK. I've tried the possible solution to create separate inbound and outbound trunks for each DID (shown elsewhere); this did not work. [Note: this is not a license issue; 'HC' confirms that there is no limit on any version for for number of SIP trunks.] This is critical for use! PBX is useless if this doesn't work. I can post logs for a specific test if you specify one. Dave