pythonpoole

I replied to your PM.

I don't believe IVM supports video playback as of yet. Express Talk does have some SIP video functionality but only in the sense that once a call is connected you can use a webcam to broadcast video to the other party. It is not built to automatically answer and broadcast on demand for incoming calls.

1) You'll need more than just a basic license sine that license does not permit any OGMs. The OGMs (out-going-messages) are what drive your call-flow applications. They play recordings, collect and process user input. Without any OGMs, your system would just be an answering machine. You'll need a bare minimum of the small business license, still with only 3 OGMs, there is not much room to play with. It is possible for you to use one IVM installation for a system of operators however, so you can do that to save on costs and upgrade to one of the higher Interactive or Enterprise licenses. 2) There are some professional telephony boards that support 4, 8 or even 16 lines on a single PC (usually a maximum of 4 lines per card installed on the PC). Compared to voice modems, telephony boards are much more expensive however. You are correct that most voice modems only support 1 line. Voice Modem / Telephony Board Comparison 1 Line per card / up to 4 lines per card Average voice quality / Better voice quality Suitable for answering machines / Can be used for all sorts of applications including integration with a PBX for call routing and transfers 3) If you are using a digital PBX such as Axon or Asterisk, you'll need to connect the phone lines using either a) Telephony Board FXO Adapter A Voice Modem would not be suitable for this application. Telephony boards are discussed above. FXO adapters essentially convert your analogue line into a digital VoIP line that can interface with IVM and most digital IP PBX systems. FXO adapters are not too expensive however usually only support 1 line per device. Each FXO adapter requires a free Ethernet port on your network and requires some time to configure properly.

Are you sure the modem is actually hang-ing up? If it is, there seems to be some kind of TAPI driver communication problem that is preventing IVM from recognizing the disconnect. If the modem is actually disconnecting, perhaps the regional disconnect tone settings are not correct for your area.

You can enter any characters you want into the log field. For example once I wanted to make a HTML log file that you could view in a browser. Basically I just created a file called log.html, added <table> to the top of the file and saved it. In IVM I just used a log string like <tr><td>label 1</td><td>%var1%</td></tr><tr><td>label 2</td><td>%var2%</td></tr><tr><td>label 3</td><td>%var3%</td></tr> The result is an HTML page that displays the log information in a table. You could for example use <br> to enter a new line in an HTML based log file.

Good to know. Is this likely to fix similar problems with other providers?

NCH is working on the solution. It appears to be a combination of XP SP3 with telephony boards.

Are you sure Magic Jack lets you use other SIP clients/soft phones, I'm pretty sure the account is locked to the device. Part of the SIP protocol includes a message that is sent from the client to the server that says what software and version the client is running. Magic Jack could easily configure their service to block all clients that don't send the Magic Jack client info. I don't know whether they are employing this technique, but it seems plausible since they seem to force you to use their hardware. There are thousands of other SIP services out there that are 'open' allowing you to use whatever client software or IP phones you wish. Edit: I should add that the best way to connect IVM to a locked device is to use an FXO adapter so it goes MJ adapter -> FXO adapter -> (Axon) -> IVM. The FXO adapter takes in an analog line (such as the one you plug your telephone into on the MJ device) and then converts it to an (open) SIP line you can connect to with PBX software such as Axon or Asterisk, or you may be able to connect directly to Axon.

It's probably a false-positive (anti-virus mistakenly thought whatever IVM was doing was suspicious). You will probably have to manually adjust the settings in your anti-virus software to allow IVM to run.

In the Axon forum you will find a detailed guide (sticky topic by me) for setting it up. The guide is actually for Axon which is designed to sit between the FXO (the adapter) and the FXS (Express Talk). I assume you want ot use the FXO to connect to a physical line from your local telco to handle inbound/outbound calls (what an FXO is for)? The linksys FXO adapters usually also have 1 or 2 FXS ports which you can set-up as extensions. The instructions aren't that simple though. The Linksys by Cisco products like the 3102 are business class networking devices that are quite sophisticated and there are literally hundreds of configurable options which makes it highly adaptable in different environements, but also a difficult to set-up for the average user.

You will have to build your own (or have someone build you a) plugin to accomplish this task, I don't think any of the sample plug-ins will do.

Using a plug-in, you could have it keep adding to a variable and send it back to IVM. If it exceeds the number you specify, the OGM can then be set to go to another OGM that says something to the effect of 'sorry, we are not accepting calls right now'. The plug-in would have to take care of resetting the variable on a daily basis.

You can use a variable to keep track of how many inputs have been made. Along with a plug-in, keeping track of information is pretty simple. I would say however that NCH should consider adding this in as a built-in feature. It is standard in the telephony world to have a limit on the number of inputs, and most other software I have used for telephony applications is able to automatically keep track of inputs and redirect them on errors.

Do you have an administrator account on the pc, and is there any software which may be blocking the services from running in the background?

Apart from trying to read the status through the event logs or the status window in ET, I don't think there is an easy way to go about doing this.

I haven't used IVM's Voice Mail feature for a while now, but last time I did you were simply asked for the pin/password if I remember correctly, so you didn't have to put in the ID at all. Perhaps it has changed since then.

No additional hardware (such as modems, telephony boards etc.) is required for VoIP lines. Only the PC and Internet connection are necessary to support multiple phone calls.

I haven't worked with passing variables in a while, so I can't remember if that is the right syntax. I do however seem to remember that you need to have the TTS in another OGM (i.e. have an OGM that executes a plugin and grabs the order and then go to another OGM that reads it. I'm not sure if this is the problem in this case.

1. No, I can't think of a way to do that with IVM. Your program would have to manage the times and send the call to IVM when ready. IVM has some basic control however, for example you can say only make calls between 10 am and 6 pm and it will only put through calls in that time-block but no specific date/times can be selected and certainly not for individual calls. 2. IVM will keep a queue going of all the calls that need to be processed, and as a line becomes free the next number in line will get called.

That's a bit strange. By analog to digital, do you mean PSTN to VoIP or are you still connecting to a physical phone line through some sort of TAPI compliant device? You can always try re-installing, just make sure you re-install using the same installer (version) that you have purchased the license for.

Is there anything in the log indicating IVM is trying to answer and play the message? Have you checked the number of rings before IVM is set to answer, and also make sure you don't have any time based routing turned on.

For the first two weeks there will be no restrictions (promotional trial), after that only the first 3 lines and 10 extensions will work.

There should be 2-3 echo cancellation options in the PSTN section of the 3102 config page. If you set them all to yes, there should be a noticeable difference (virtually no echo). You may also do the same for the line 1 options, although I think those are enabled by default and really only apply for internal calls between the 3102 and Axon.

One possibility is to use a VoIP provider that has failover options. Basically this means if the call can not be put through to your Axon box for whatever reason (e.g. system down time, power failure, Internet disruption, etc.), calls can be forwarded to another number instead. One provider that offers this feature free of charge is Voip.ms however last time I tried it didn't work so well with Axon (although I think NCH may have corrected this in the latest version). Voip.ms offers virtually unlimited incoming for local Australian numbers at a rate of $10 USD/month which handles up to 2 simultaneous calls. They also offer Australian toll-free 1800 numbers for $10/month plus 6 cents/min and they handle up to 50 simultaneous calls At this time there is no porting option for Australian numbers however. I'm sure there are probably other Aussie providers with failover, I just don't know which ones. You can try Faktortel, they seem catered towards business users and they may have some failover options.

I don't think this is possible. It would require Axon to send a hook-flash command to the 3102 and I don't think there is any way in Express Talk to signal that you want Axon to hookflash. What I have done in the past is sign-up with the telco's 'Forward on Busy' feature instead of call waiting and then either have busy calls go to a mobile phone, or better yet to a VoIP number that is linked with Axon (that way you can share the benefits of having a PSTN landline (reliability, security, performance), and the benefits of a VoIP line (cost saving, allowance of simultaneous calls) on one system).