pythonpoole

This is because the audio is not being generated by Text To Speech, instead it is playing pre-recorded prompts for digits/numbers and words such as "dollars" and "cents". You can record your own prompts (replacing the existing ones) by clicking Settings and then going to the Prompts tab.

You are automatically entitled to a free upgrade to any version released within 3 months of the version you purchased. The programmers may also decide to let you upgrade free of charge beyond 3 months if the update is pretty minor (e.g. bug fixes, maybe a new feature here or there) Beyond that you can upgrade your license to the latest version at a discounted rate using this form http://www.nch.com.au/upgrade/ Each time you upgrade, the software is a fresh install and you will not need any previous versions to run it. My understanding is if you download and run a newer version that is covered by a free upgrade, it should automatically detect your existing product activation and activate the software for you (although I'm not entirely sure on this one). If it still asks for you to activate the new software, try entering your registration details. If it says the details are no longer valid for the upgraded version, then the software is not covered under the free upgrade policy. You can then re-install the version you purchased (4.0) to continue using a licensed copy of IVM.

There is a section of the IVM manual which refers to delays in detecting DTMF key presses. However the manual makes it sound as though its more for recordings, but you can always try and see if it helps. It seems however that the delay may actually be caused by your modem hardware and not necessarily by the IVM software.

This is most likely the result of a regional incompatibility. Even though there have been many efforts to standardize communications across the globe, each region of the world tends to do things just a bit differently to everyone else, which leads to problems. Where are you located? If for example you imported a voice modem manufactured for the North American telecom system, and then attempt to use the device in Britain for example, the device may fail. The reason is, many different regions use different tones and ways of signalling such events such as hang-up,busy,Caller ID, etc. This means modems that are not specifically built to be compatible with the system standard used in your area may have trouble detecting some of these events, and therefore it may result in recordings of dial tones and operator instructions as you point out. To give you a specific example.. as I understand it, the North American Bellcore system leaves the phone line completely dead when all phones are on the hook and it's not in use. In countries like Australia however, I believe certain pulses of tones are sent down the line when it is not in use to let the devices know on the other end it is still 'active' (i.e. physically connected to a phone service) but is simply not being used at that particular moment. When you use equipment built for North America on a line that is sending tones to keep it active, the equipment could easily misinterpret this as actual call activity and therefore fail to hang-up. I suggest checking with your manufacturer to see if they can provide alternative drivers/firmware for your region. Edit: Also you should check the End of Call settings in the line's properties window, here is a quote from the IVM manual:

1) There is no difference between the trial version and full version other than the former expiring after about 14 days If you need any help, I am here pretty much here every day to answer most questions you may have about IVM, Axon, and Express Talk. Also, what you're trying to accomplish is pretty straight-forward and shouldn't be too difficult to get working. 2) This forum is the best place to obtain support. In most cases, I will answer questions within 12 hours, sometimes as quickly as a couple of minutes. If I do not know the answer to a question, I will leave the topic open for a bit and give someone else the chance to answer. If after some time nobody else is able to provide an answer, I often post a reply with links to contact NCH Support directly. NCH does have a technical support system where you can submit your question and they will respond by e-mail. The problem is if you haven't specifically payed for one of the technical support packages, your support ticked is treated with low priority and in many cases you will receive an answer much faster using the forum. If you have purchased the software you are asking technical support about, you will be put higher up on the priority queue than non-paying customers, but will still have low priority compared to users who payed specifically for the support service.

I'm sorry I can't be of more help, I really have no answer for you. Have you tried contacting NCH directly? Technical Support Contact Page: http://www.nch.com.au/support/supportconta....html&pty=7 Bug Report page: http://www.nch.com.au/software/bug.html?software=Talk Software suggestions page: http://www.nch.com.au/suggestions/index.html?software=Talk You can also try to contact CISCO about the problem, they at least may have some suggestions or may be able to identify what exactly is causing the problem. Lastly, you can kindly PM nchaj (software support manager), who will likely pass on the details of the problem directly to the program developers so they can fix the problem in the next upcoming release.

Setting up the HT503 should actually be a lot easier than an SPA3102. Assuming the HT503 is on the same network as the Axon computer, you should be able to simply fill in the 3 details below in thr HT503 configuration page after creating an extension for the device in Axon's configuration page (assuming you are using it as an FXS device) 1. SIP server and/or Outbound Proxy FQDN or IP address Set to the IP address of the Axon computer 2. User ID, Authentication ID Set to username / extension number in Axon (User and Authentication should be the same) 3. Password Set to password in extension Save the configuration and reboot.

It's a bit strange if it only works on PPC considering this product is primarily built to run on Intel based PCs (meaning it'd be a lot easier for them to port it to Intel Macs). If indeed it only runs on PPC, I would have to take a guess and say the Mac version probably a bit old and has a low demand, thus is not being actively worked on to resolve bugs and provide updates. I don't have a mac, so I can't really test the software.. however I do have an OS X Intel vmware emulator and every time I try and run Express Talk it takes up 100% CPU and I literally can't do anything (I can see the GUI but the app is basically frozen solid).

You could probably accomplish this with either the GETDATA, GETDATAX or SQLLINK plugins on this page http://www.nch.com.au/ivm/plugins.html However it may be better to custom build a plug-in that is better suited for what you're trying to accomplish.

I think the best way is to use a plug-in with IVM. The plug-in would reference a database to see which ring group is associated with that extension, and then it would return this to IVM as a variable which IVM could use in the next OGM to transfer to. E.g. Internal employee dials 105. Extension 105 rings. Outside caller calls in and enters extension 105 (or some kind of reference number like employee id). IVM references the database and determines that 702 is the correct ring group for that number. IVM transfers to 702, and extension 105 along with a couple of other extensions ring.

IVM already allows you to set a password for each mailbox. Under the mailbox properties window, select the Delivery tab and enter a pass code (typically 4 digits) for that mailbox. Then make sure you set the OGM that users call into on 199 is set to 'Access Mailbox Control' and most importantly that the 'Prompt for Mailbox Access' option is selected. Currently there is no way for the user to change the pass code from their phone, but the Admin can issue random passwords to users and change them if needed without seeing what is set. Note: Each mailbox must have a unique password. My understanding is it accesses the mailbox based on which pass code was entered. The advantage is users can pick-up their messages from other locations, their messages are always password protected and its convenient to pick the messages up. The disadvantage is if users/admin select the same pass code for 2 mailboxes it could cause conflicts, also since it doesn't prompt for a mailbox and password, it may be easier for people to guess (or accidentally) type in a different pass code and get access to someone else's voice mail.

Sorry for the misunderstanding, I'm having trouble figuring out what you're trying to accomplish. In axon you can select the group or extension you want each external line to ring for an incoming call. I assume by 'I do not want the main call to RING on a group of numbers' that you will set it to ring a particular extension. This is the part that confuses me, 'Can that extension dial a group after being routed to the extension?' and 'just the extension ring a group after being routed to that ext.' Isn't this the same as simply routing to a group in the first place? If I can better understand what you are trying to do, I'm sure I can come up with a solution for you, but at the moment I am at a loss to the purpose of all this. Is it that you want to be able to call in an extension directly, and then let callers dial an extension which actually dials a group of extensions? Are you trying to differentiate between calls made directly to an extension vs calls sent there (e.g. by an IVM transfer)?? Please provide further details.

Well for one thing, the group number needs to be different to the extension number (how else would Axon be able to tell whether you wanted to call just 1 extension or the whole lot). Typically group numbers start with 7, and extensions with 1. However if most or all of the time you want to ring more than 1 extension when calling in, you could choose to switch this so that by default when you call 105, it calls the group of extensions related to 105 including 105, and when you call 705 it rings the sole extension directly. I'm not exactly sure what determines which voice mail is used by default when there are multiple extensions in a group. I would think (from a programmer's perspective) that it's either: 1) The mailbox of the first extension in the group list 2) The extension which times out first (has the shortest cut-off before voice mail kicks in) If you can't get it to go to the right mailbox, you can use IVM's confirmed transfer feature. If the call transfer is not confirmed (i.e. not picked up in time), you could set it to go to a specific extension's mailbox.

I answered your post in the IVM forum. Next time, please try to stick to one forum... it keeps things organized and helps keep related discussion in the same topic rather than across different forums.

Yes, when using IVM in conjunction with Axon, a ring group/queue can be created and used to ring multiple internal extensions at a time.

I assume this is the problem mynumber@neofon.tp.pl@neofon.tp.pl In the username/SIP umber field you should just have the number (mynumber) and not the IP/Host name (@neofon.tp.pl), otherwise Express Talk will try to register username "mynumber@nefon.tp.pl" with the "neofon.tp.pl" when your username is only "mynumber"

This is strange, I've never had this problem. What voices are you trying to use? Are they SAPI 4 or 5 and are they the 'natural' voices (e.g. at&t, loquendo, realspeak/nuance) or the lower-end voices (e.g. Microsoft/L&H)

You'll need a plug-in for this. The plug-in can return a string containing one ore more variables and their corresponding values. If the plug-in returns a nextogm variable, IVM will use this to direct the call to another OGM determined by the plug-in. You could also return a variable contain the person's name. Then it's simply a case of having nextogm set to an OGM that transfers the call to an operator if no name is found, and if a name is found to an OGM where a text to speech script exists that will read out the user's name in a sentence (e.g. Welcome back %uname%)

It depends on the phone you're using. Unlike analogue telephony, VoIP phones don't use tones or flash hooks to signal switch lines, instead the phone manufacturer will develop their own method to switch between phone lines and then use the standard SIP protocol to relay this information to Axon / the PBX. For some IP phones and soft phones there are 'line appearance' buttons which allow you to switch between virtual 'lines' as needed (e.g. when a call waiting call comes in, the 2nd line appearance button flashes, and user can switch to the call waiting call by pressing the flashing button). On other devices, the old 'flash hook' (hanging-up and quickly picking up again) can trigger the device to relay a line switch to Axon, or sometimes there will be a clearly labelled 'Flash' or 'Link' button on the phone itself you can use to switch calls. Note 1: Not all IP phones or devices support multiple lines/call waiting (in VoIP telephony, Call Waiting is the same as having a 2nd line appearance for incoming calls) Note 2: Some IP devices need to have their Call Waiting feature enabled before it can be used

If you're using IVM to transfer calls to the extension number they put in, you can limit the accepted data values (e.g. from 100 to 110) and any numbers outside of that range will automatically trigger IVM to say the number was invalid and to try again (repeating the OGM)

IVM is great for pretty much any IVR system. You may however need to use a plug-in in order to connect your product database together with the caller's input. The basic concept is, create an OGM that is in data entry mode. You can then configure the OGM for how the data should be accepted. Example: if the number of digits will be the same for each item, you could limit the number of digits to a certain amount so the user's input is automatically accepted when the full number is entered. Alternatively you could let the # key signal the end of the data entry. Other settings are also available such as limiting the accepted data values so if the caller enters an invalid/incorrectly formatted number, the caller is then told and asked to re-enter the item's number. After the data entry is accepted, it should be stored to a variable (e.g. %calleritem%). The OGM should then take the user to another OGM which will determine what to do with the input (the value stored in %calleritem%). A plug-in could be used for example to cross-check a database and retrieve the next OGM the caller should be taken to (good for a small system with a limited number of items). Alternatively, the Database plug-in could retrieve information about the item and send that back to IVM as a variable, IVM could then be instructed to go to a new OGM which reads aloud the information using a TTS voice (or using a recording if you can adjust the plug-in to modify the OGM in real time with the new audio file). If you have any further questions, please don't hesitate to ask. For new users, I recommend taking a look at the IVR and plug-in example files here for help on getting started: http://www.nch.com.au/ivm/examples.html and http://www.nch.com.au/ivm/plugins.html

Yes, both the P4K and T4K Yealink models should work properly with Express Talk. Some of Yealink's other phones also work (simply because the driver and functions happen to be similar, meaning it's mostly just coincidental when it does work). Note however that even if you take the risk and get another one of Yealink's phones to work, there is a high chance that some feature(s) won't work properly (e.g. caller ID). Note there are several other wireless IP/VoIP phones out there that work independently of your computer / the software you're using. These phones connect to a nearby WiFi connection and can be placed anywhere within range and not have to be connected to a PC, Ethernet cable, or any other linking device to work. These phones are sometimes quite expensive however and can rack up $200 in costs depending on the manufacturer.

I'm not sure this phone is supported. The manufacturer of the phone claims only support for Skype, and NCH only claims support for the Yealink USB desktop phone. You might still be able to use the phone as a headset, but it's unlikely you'll be able to get it to ring, dial out, or have any of the features on the phone work (e.g. digit presses, caller ID).

Yes, if you are using some kind of plug-in or the built-in LOG feature to store the info in a DB, by the time the OGM ends (when the action such as leave voice mail is performed), the details will already be written to the DB/CVS. Make sure the recording for that OGM is at least 1 or 2 seconds long though just to make sure the plug-in had time to execute and such.

Yes. You can have the call answered by any OGM you wish and then after storing the person's name and address to a log/database, use the OGM action to send them to Voice Mail to leave a message.