pythonpoole

At this point in time, the actual 'OGM message audio' has to be pre-defined/loaded into the OGM properties window manually. You can then use the SDK/API command-line argument "ogm" to specify which OGM to answer calls with (thus changing the audio on demand). See example below: IVM.exe -ogm mysecondOGM However there is also another solution... Under: \Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM\OGMs, You should find 3 files for each OGM you have. 1) An .ini file which contains configuration information for the OGM 2) A .wav file with the OGM message audio 3) A .txt file with the script/notes for that OGM To dynamically change the OGM's audio during run time: METHOD 1: Add/replace this line in the .ini file (under the [iVR] category) to make the OGM play an audio file of your choice: playexfile=C:\mysound.wav *Note the OGM message should be blank (e.g. 1 second of silence) as this audio file will be played in addition to the normal OGM audio METHOD 2: Replace the .wav file in the "\Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM\OGMs" directory with a new one To add/modify the OGM's Text To Speech script at runtime: Modify the .txt file with the new TTS script *Make sure 'Text to Speech Voice Synthesis of Script' is checked in the Advanced tab of the OGM properties window before hand For your second question, yes you can use a plug-in. Additionally you can use the built-in "Add Entry to Special Log" feature to enter log info every time a OGM runs. By having each OGM add a line to the log with the Caller ID / date & time you can track a call through each OGM and therefore see what options were selected.

It seems strange to me that SJPhone's transfer feature works and not Express Talk's... they should both be sending the same SIP transfer request. When you turn the log/status window in Express Talk on, does it indicate it is initiating a transfer? Also, are you sure you are using the Transfer feature correctly? I believe the procedure is as follows: 1) When a call is in progress, press the transfer button 2) Type in or select a number from the drop-down list to transfer to 3) Click Dial When the call has been answered.... 4) Click the Transfer arrow (beside the Dial button) to finalize the transfer The last step is not that clear and is easily skipped over. I suggest you try again using the 4 steps above and see if you can get it working this time.

I think nchgb is on to something. I believe you said "BT services on the line" which would indicate to me if I'm not mistaken you are in the UK. If the board is from a US manufacturer, or was purchased overseas, it is highly likely the voice card is having trouble detecting things like dialtone, ringing or Caller ID since these are very region specific and differ significantly in the UK. The best thing to do is to update the drivers as nchgb suggested and see if you can find any information on changing the regional settings for the board. You can also download a TAPI diagnostics program (quite a few available, just use google) to determine whether it is the card that is unable to detect the ringing and so forth or if somehow the Card isn't communicating with IVM properly and relaying the information.

IVM works with almost all Windows Operating Systems (Windows 95/NT4/98/2000/Me/XP/2003/Vista) Are you referring to Windows Server 2003, or the newer Vista based server? I can't guarantee compatibility with the Vista server, but it should definitely work fine on 2003.

In Carousel go to Settings -> Telephony (tab) -> Add... and if the Synway board is detected, it should appear in the list. If so, it should be compatible with the software, just click Add again to use the device with Carousel.

The server address should be the IP of the computer with Axon on it (you can use localhost or 127.0.0.1 if it's the same computer). If you go to "Getting Started with Carousel (using the Carousel Wizard)" in the Help file, for every option it will tell you what you should enter if you are using Carousel with Axon. Axon includes a pre-configured Carousel line by default and if Carousel is set-up properly, it should interface with Axon's carousel line without problems. Then you can select where incoming calls are routed to and how outbound calls should be handled (through a dialling plan). I am curious to know whether Carousel was able to find and connect to your telephony board.

If it's a CAHTA, CURL, or Synway board, you can use NCH's carousel software as a 'virtual FXO adaptor' (free of charge, acts just like a normal FXO device, but only supports some boards) see http://www.nch.com.au/carousel/index.html#102 Also the link you provided (http://nch.invisionzone.com/index.php?showtopic=5091) is exactly the guide you asked about (for setting up a SPA3000/3102 with Axon). The guide was also written by me

Ok, it wasn't very clear in your post, but are you using some kind of FXO device? Vonage has decided to lock all their devices so they can't be used with other services or software like Axon. This means, you are left with only 2 options: 1) Purchase Vonage's Softphone plan (this opens up your Vonage account so Axon can connect directly to it easily without any additional hardware) PRO: Easy CON: Costs involved, order in addition to normal Vonage plan) OR 2) Using an FXO adaptor such as the SPA3000 OR SPA3102 PRO: No extra ongoing costs CON: Purchase cost, more difficult to set-up (this means your Vonage VoIP line will be converted to a PSTN analogue line from the adaptor they gave you, and then sent to the FXO device which will then convert the analogue line back to a digital VoIP line and send it to Axon. Then you may have analogue phones you wish to connect to Axon with ATA adaptors which convert digital VoIP lines to analogue phone lines. A lot of unnecessary conversions, just because Vonage doesn't want you using other non-vonage hardware/software).

I'm unable to determine why there is a problem. As far as I can see IVM receives the command to go to the r11 OGM and then it performs as expected by using "Command - Go" to go to OGM r11. Unfortunately it disconnects right away which is a bit strange. Here are a few questions to help solve the problem: - What action is the r11 set to take when the OGM times out (i.e. "Then:")? - When you preview the Audio in the OGM by pressing "Play Message", does it play? - What happens if you change the audio (e.g. to text to speech) and try again? - If repeat is set to 0, try setting it to 10 and see if it changes anything. Lastly, the number of OGMs you can have is limited by your license: - Small Business License - 3 OGMs - Small Interactive System - 70 OGMs - Professional License - 70 OGMs - Enterprise License - Unlimited

For your first problem, I think this may have to do with your VoIP provider. The provider should signal back to Express Talk if a number is unavailable/disconnected and Express Talk should then play its universal busy/error tone and report the error in the status window. Also, you are correct, Express Talk will play its own ring-out tone until the other party actually answers the call (ignoring error announcements and third party ringing), and so far there is no way of avoiding this.

It appears that when you upgraded, it only gave you a professional license. I suggest you contact NCH about the problem. The problem is NCH's current policy generally refuses refunds in the event that: - The software has already been activated - A mistake was made during order/upgrade process (e.g. selected wrong license) If in fact it was a case of accidentally selecting to upgrade to the professional license instead of the enterprise license, you may be unable to obtain a refund. However, I think they may let you pay the difference in price and upgrade to the Enterprise license to fix the situation. Of course if it's there mistake, I'm sure they'll help rectify the situation as soon as possible.

I don't believe you can do this. The best thing to do is keep the message short (e.g. 'Please enter your pin' or 'Please enter your credit card number'). By doing this, most callers will wait until the end of the message anyway before inputting a data string. If the message is too long, callers could 1) become confused and 2) start inputing while the message is still going on. The second thing to do is to determine an appropriate repeat value. If users are only expected to enter a pin number (3-5 digits), then the repeat time could be as low as 8 seconds. If the entry is 6-8 digits, around 16 seconds is more appropriate. For larger numbers (e.g. credit cards), boost the time to around 25-30 seconds. Note that entries where callers are expected to know the value to enter off by heart (e.g. 'Enter your phone number', 'Enter your zip code', 'Enter your pin') can have slightly faster time-outs to entries that require callers to refer to a piece of paper for example (e.g. 'Enter your account number', 'Enter your credit card number', expect callers to use an extra 3 or more seconds to find it on paper/card, remember the first set of digits, enter that, next set of digits, enter that etc.). Always make sure the * key allows the caller to go back, or at least repeat the instructions should they forget or not hear correctly.

I think you mentioned you were using this modem with a PBX system in another topic. Is it possible the dial tone used by the PBX is (even a little bit) different to the standard North American dial tone (or whatever tone the modem is expecting)? I know that dial-tones are very region-specific and can even vary between phone devices like PBX systems and ATA VoIP adaptors. If the tone is the wrong frequency or varies in pitch instead of being steady, it could cause the modem to incorrectly asses that no dial tone is available for calling. Also, (I don't have a link for this, sorry) there are several TAPI diagnostic programs online to help debug problems you're having with TAPI compliant modems. They basically show a log of every TAPI command the modem executes. If the log reports no dial tone, then it indicates the problem is indeed with the modem detecting the dial tone, and not IVM. If it does report a dial tone and IVM does not, it indicates a communication problem between the modem and IVM.

I've had the same problem for a while now. I'm using 'virtual webcams' (i.e. those virtual webcam drivers that are installed with webcam effects programs). Eyeline continues to crash unexpectedly with abnormal exceptions, and in that sense it seems a bit unreliable. The crashes appear to be random, and I can't find much of a pattern to them, but they happen often enough I can't just leave it running without expecting to come back to it crashed.

It could be the modem drivers then. I know some users have reported certain driver versions causing incompatibilities with IVM, I'm not sure if that applies to your modem. Try and see if you can download the latest driver from the manufacturer's website.

Well the Linksys SPA941 is the only phone NCH specifically recommends here: http://www.nch.com.au/hardware/ipphones.html I would say that pretty much any SIP based IP phone will be compatible. Just remember though that Axon won't support all the features that your phone may have buttons for. For example, Axon provides no built-in call parking system. So what I mean to say is, you should be safe with most SIP telephony hardware (so long as it supports standard codecs like G.711 ulaw/alaw or GSM), just keep in mind that many high quality SIP phones will probably have some additional features like Shared Line appearance or visual voice mail that won't be able to function with Axon. P.S. This post suggests Polycom phones work fine: http://nch.invisionzone.com/index.php?s=&a...ost&p=13025

The first issue could be the result of incorrect regional settings. For example, the modem may be expecting a North American dial tone, and instead is getting an Australian dial tone (as an example) and it is unable to detect it (or vice versa). Some modems may offer a way to change regional settings to work in other countries around the world. The second possible cause is an incompatibility. The modem may not be fully TAPI compliant, and perhaps is not properly signalling to IVM that a dial tone has been detected. For your second problem, I am not sure. Sometimes your phone company may require a special dial string for transferring calls (different to the default !,) and it must be changed in order for it to function correctly. Try a blind transfer and see if that works. If it does, I'm inclined to think either the phone service does not support confirmed transfers, or the dial string for confirmed transfers is different. Edit: Nevermind, I just saw your other topic and you specifically noted that you were using Vonage and the correct dial string codes. I don't use Vonage any more so I can't test confirmed transfers, and I'm not sure what the success rate is for other Vonage users.

4.01 is a very obsolete version, since then several new releases with bug fixes and additional features have been made available. Perhaps you should consider upgrading to the latest version (4.08) with special discount pricing using this link: http://nch.com.au/upgrade/ I'm not sure how many people have 4.01 lying around though. Your best bet is to try and find an old shareware site that happens to have a copy of it still up for grabs. Otherwise, I guess you can wait around for bit here and see if anyone else has it.

This isn't going to be any help, but I thought I may as well point out that I also tried to get DNS working with IVM, but eventually gave up. This was a very long time ago, but I vaguely remember doing some diagnostic testing and discovering it had something to do with IVM giving the wrong extensions or file formats to the audio recordings. When it did this, DNS was unable to process the file for speech recognition and would error out with something like 'Invalid file format' (error not visible). I discovered that if I manually corrected the file type (I think I may have used some audio conversion utility), and then got DNS to process the file, it worked well ...as well as a speech recognition engine usually works, which isn't too great for phone recordings. I could never get the process to be automatic though.

Well you're right Axon doesn't support this feature natively, so I've decided to put my thinking cap on until I can come up with a solution. ... Ok I thought for a while, and I think there may be a way to simulate a call parking system using IVM. If it works, you have to understand it will be a pretty limited in functionality, but at least it should work. My idea is, have an extension set-up with IVM which goes to an OGM that runs a plug-in which will store a variable (e.g. callParked) and set it to true for future use. Then the OGM will be set to go to another OGM that plays a short 'on hold' music loop or message and at the same time will run a plug-in and checks for a NextOGM return value. This OGM should be set to repeat itself indefinitely, and should only be about 10 seconds long or less (you want it to keep checking in with the plug-in). The plug-in will need to store some data globally (e.g. to a text file). The basic idea is the plug-in will keep track of 2 variables (e.g. callParked and pickupExt). When a call comes in for the first time (to the 'intro' OGM) the plug-in will set the callParked variable to true. Then in the next repeating OGM, each time the plug-in runs, it will check to see if the callParked variable is true. If it is true, it should return something irrelevant (e.g. setting an unused variable to a value) so it doesn't interfere, and the OGM will continue to repeat itself. If however the plug-in checks and callParked is false, then it will return NextOGM with value 'parkcallback' (see below) and a global variable (for IVM) called pickupExt. Now there should be another extension ('2nd extension') set-up in IVM. When someone calls into this 2nd extension, it should run the plug-in and set callParked to false and then pickupExt to the cid (caller ID) variable (this will tell the plug-in who phoned into the 'unpark' extension (2nd extension). Now the 'parkcallback' OGM will be set to transfer the call to the global pickupExt variable that was set by the 2nd IVM extension's OGM, so that when a caller calls into the 2nd IVM extension and then hangs-up, IVM will send the parked call to the parkcallback OGM which will then call back the person who just called the 2nd IVM extension and let them speak to the previously parked caller. _______ Ok, you probably didn't understand a word I just said.. After re-reading I realize I just kept rambling on. But here is a summary of how such a system should perform: Pros: - You can park a call by transferring it to an IVM extension - While the call is parked, you can play a small music loop/message (I wouldn't use anything >10 sec though) - You can call into another IVM extension to let IVM know to unpark the call (you call in, then hang-up, then IVM will transfer the parked call to your extension) - You can pick-up/unpark the call from any extension, not just where you parked it - You can set-up a time-out for the OGM to have IVM automatically transfer back a parked call if it is left abandoned too long Cons: - Limited to 1 parked call at a time (you could have more by re-creating the same system twice, but you would have to send the second caller to the second parking system if the first one is use, and you wouldn't be able to tell if it was). - Limited MOH (music on hold) capabilities (you won't be able to use IMS, and the music/message has to be short enough that the plug-in is checked often i.e. <10 sec, if you use music (recommended) it should be a 8-10 second loop) - Unparking isn't as convenient as a native parking system. You'll have to phone in to an extension, hang-up and wait up to 10 seconds (as long as the OGM message is) for the unpark call to come in. Despite the cons, it is theoretically possible to set-up and I can't see any reason why it wouldn't be possible to put it into practise.

I don't know much about the Cisco Call Manager / PBX. But I do know that some PBX systems (e.g. Asterisk/FreePBX) have "Feature Code" settings which let you specify DTMF tones that will automatically trigger a feature such as call transfer or call recording. This means that even if the SIP phone, soft phone, or ATA adaptor couldn't initiate a transfer using its built-in feature, by dialling a feature code (e.g. ##) in call, the PBX will automatically set-up a call transfer. I have no idea if and how that would work with Cisco Call Manager though.

Well I have good and bad news. The good news is, there is something called the 'Unified API' NCH has developed to help developers have more control over many of NCH's applications. Axon is shown as one of the compatible applications for this, and the API web page goes all the way to give this example: "What if you want to setup an extension in Axon for a specific time? Simply using the Windows Scheduler and the Unified API to send a command to Axon informing it to setup the appropriate extension." The other good news is the API kit has pre-written examples for all sorts of programming languages, including VB. The bad news is, I cannot find a single reference to the API on the Axon page, nor can I find any documentation on what commands are available for controlling Axon. The Unified API page says that there should be documentation on how to use the API on the web page for that software, but Axon's page has no such link.

It's up and running right now actually, it's just that it's not available to the public just yet until I finish testing with IVM/Axon/Express Talk. If you're willing to a be a beta tester though I can set-up an account for you and add a couple of dollars of spending money to test outbound calls and transfers. I'll PM you with the details in a bit.

I don't know if you've already seen this topic, but it looks like it should help you out a bit http://nch.invisionzone.com/lofiversion/index.php/t3833.html Personally I have never used IMS with Linksys devices, so I can't really offer much help. It seems to me though (from the topic) that the people who were most successful put in the IMS extension @ the IP address of the MOH (IMS) server : followed by the port number IMS is using (e.g. 197@192.168.0.3:5070).

First things first Axon: is used for routing and managing calls on your phone system. For instance you can have multiple extensions connected to Axon and have incoming calls ring a certain number of extensions. Meanwhile each extension can call each other, or make an outbound call while other calls are already in progress. You can also control the outbound dial plan which essentially determines what line to use and how to dial the number based on what the person dialled. IVM: The basic use of this software is as an answering machine for VoIP calls, Voice Modem calls, or calls from Axon. It's other main use is as an Interactive Voice Response (IVR) application. It will allow you to create OGMs (out going messages) with menus (e.g. Press 1 for Sales, 2 for Customer Service), as well as data input, and the ability to interface and process information using plug-ins (e.g. to accept credit card payments over the phone, or read back the weather forecast for the caller's area). Softphone (Express Talk): Used just like a normal phone accept the line is connected to a VoIP server (or software like Axon) which allows you to make calls over your network or Internet connection. You can also use an IP phone in place of a softphone. IP phones look, feel and act just like a classic PSTN phone, but communicate over your network like a softphone. Last but not least, you can get an ATA adaptor which will convert your normal PSTN phone into a fully functional IP phone. I'm not exactly sure what you mean, could you provide more information on what you're trying to accomplish? Streaming voice over dial-up is not recommended since there is very limited bandwidth available.