pythonpoole

You will need an FXO adapter to interface analogue lines with Axon (IVM does not need such an adapter). However, some Telephony boards are compatible with NCH's Carousel software which acts as an FXO adapter and converts analogue lines connected to certain compatible Telephony boards to digital VoIP lines for use in Axon. The Sipura 3000/3102 adapters do the same thing, but they are an all in one device (i.e. they contain all the hardware and software/firmware to convert an analogue line to a digital VoIP line for Axon to use all in one small device.

Currently the built-in voice mail mechanism is somewhat basic and there isn't really much you can do to add features to it if that is what your asking. I.e. I don't believe you can add any menu options to the 'checking voice mail' menu. I think the system needs a bit of a makeover, and frankly considering IVM is designed to be pretty much an advanced voice mail system, I find it hard to believe that it lacks many features and contains a couple of bugs (see below). For instance, when accessing voice mail by phone: - The date and time is only said if the message is from "Today" or "Yesterday" because the system is not able to say the months or week days - If the time of the message is something like 4:09 PM, it is announced as "four nine PM" - There isn't an option to announce the caller-id for messages - There is no option to skip a message - If the messages are old, there is no introduction to the message at all when calling to check.. the first message just starts right away, a bit unexpectedly - There is no separation between new and old messages (i.e. ones you have listened to/saved) It does have the mailbox setup menu (dial 8 I think) which provides the necessary functionality to record away messages and set passwords and what not. But that is pretty much how the voice mail system operates, and you can't really alter it. If you're looking into a more advanced VM system, you will need to program your own one using several OGMs and the various record actions available.

Basically the Stun servers are just set-up so VoIP phones/softphones can retrieve the IP address of the computer/device they're running from and what ports might be available for making connections. These are some other stun servers, no one server provides and real benefit over the other unless 1 is closer to you in which case it may provide quicker data transfer rates. # stun.ekiga.net # stun.fwdnet.net # stun.ideasip.com # stun01.sipphone.com # stun.softjoys.com # stun.voipbuster.com # stun.voxgratia.org # stun.xten.com As noted above, opening the ports used by VoIP will help heaps in terms of allowing a connection the server and also in improving voice quality during calls much like how opening specific ports enables much faster downloads on peer to peer networks.

Just to clarify: 1) Your VoIP provider will limit the number of concurrent/simultaneous calls you can have at any given time. Some providers are very strict and only offer 1. Many offer 2 channels (read as "lines") so that you can have call waiting (but technically you can use the two channels to answer both calls on 2 different phones and talk at the same time), others offer many more channels but usually at a greater cost (e.g. CallCentric I believe offers 2 or 3 channels by default, but additional channels can be purchased (at a high per channel cost monthly rate). Note that even providers that claim "unlimited" channels will often limit to 50 or something just to be on the safe side and to ensure their server doesn't get overloaded by any account holder. 2) Axon does not limit the number of incoming calls. Just because you only set-up one "External Line" in Axon does not mean it is only set-up to take one call, it can take as many as supported by your VoIP provider (see 1). The next call that comes in is processed just like the previous ones and if it isn't answered can be forwarded to your voice mail. 3) IVM is limited to the number of calls it can take (this is limited by your IVM license, then by your VoIP provider, and finally the number of simultaneous calls you allow in the settings for each telephony device or VoIP account you set-up with IVM).

Server seems to be experiencing a few problems today, ignore this post, see below.

I'm not sure what could be causing the problem, although whether or not the program did run.. I wouldn't suggest running it with only 64mb of ram. To get the minimal experience out of IVM I wouldn't use it on a computer with less than 128mb of memory, simply because phone calls take up a chunk of computer resources and unexpected errors or behaviour could result if those resources aren't available to use. Also, personally I wouldn't use less than a pentium III for the same reasons. Computer telephony integration isn't really for those old dusty computers you have no use for.. those might be useable as basic network access terminals and such.. but I would never recommend them for telephony applications if you want to ensure high voice quality and general stability. Anyway it seems strange there is no error or program in task manager (I've only heard of this happen once before, but it was for another application and on Vista not 98). I'm not sure how to help.. are you using Win 98, or 98 SE? I'm not sure if Windows 98 has the Windows System Event manager built-into it, you can check there to see if the Operating System reported any errors, warnings or unusual program activity. If not, I'm not sure what to suggest, and maybe it's best to contact NCH directly.

Firstly, paragraphs make it easier to read I highly doubt the female voice is a recording from NCH. The only recordings NCH has are using an Australian English Male voice. That recording is probably part of the firmware or driver for the telephony board or voice modem you're using. Most likely there is some sort of hardware or driver conflict that is causing an error and your hardware is programmed to automatically play that recording in such a case. Most people are able to get the product functioning correctly without much trouble at all. This seems to be a rare case of hardware/driver conflict/incompatibility of some kind. What hardware (i.e. telephony board / voice modem) are you using and is it fully TAPI compliant? Have you downloaded and installed the latest drivers for the hardware (from the manufacturer's website)

Yeah. If you have a menu set-up in the OGM that you are using to send the caller to the Voice Mail and you have ticked "Use special menu" then the menu options from that menu can be pressed during or after leaving the message.

I'm sure they'll update it soon, and yes that is the official page. Considering how fast it was released.. I would think it was just some very minor bug fixes (such as correcting the plug-in issue).

I can't find any details on whether the modem supports TAPI, but if I were you I would not take the chance unless you are sure you can return it without hassle. NCH recommends the Way2Call modem if you're looking for a high quality, portable, USB solution. Their modems are TAPI compliant and are of much higher quality than standard voice modems. They also carry a big price tag however of >$350. In that case, it may be cheaper to simply buy the USB telephony board from NCH's hardware sales partner AltoEdge at https://secure.nch.com.au/cgi-bin/hw.exe?class=TEL

Did you read my reply to your post in the SPA3000 / SPA3102 GUIDE topic?

Well I have no idea how to set-up the SPA adapter with Asterisk, I believe it is quite different to Axon (what this set-up guide is for). You can try the Voxilla forums, they have quite a few sample config guides for Asterisk and the SPA3102. P.S. I should point out that when you use this Guide with Axon, you get pretty much the same error you got with Asterisk.. the only difference is with Axon it still works even with the error.. so I don't know, is it not working at all with your Asterisk box? If there is a way to accept incoming calls without registering, perhaps try that?

If I remember correctly the setting is in the advanced External Line properties, and it's called something like "Chain to next line." Basically it lets you select another External Line to send calls to if the current one fails to place the call correctly. So if you have 4 lines, you want to chain 1 to 2, 2 to 3, 3 to 4, and 4 to 1 to ensure the call will get through just so long as any of the lines are free.

Yes this is possible, but not with a regular voice modem. This set-up would require: 1) Either a FXO adapter (such as the Linksys 3102) OR a Telephony Board 2) NCH's free Axon PBX software for your office computer 3) NCH's basic (free) or business edition Express Talk software for your laptop AND (optionally) USB Phone OR a compatible SIP Phone or ATA device. It basically works like so: The FXO adapter connects to your phone line. Axon is set-up with the FXO adapter to use it as an External phone line. (The FXO adapter converts the Analogue line to VoIP so it can be used by Axon to make/take calls over the local network or Internet). The Express Talk softphone OR other SIP hardware you are using in the remote location will then need to be configured with an Extension in Axon. After proper configuration, you will be able to use the analogue phone line at your office from any remote destination with an Internet Connection that has the appropriate ports opened on the router's Firewall.

Well I don't know about Asterisk cards, but when you use Carousel in combination with Axon, you can interface with many hardware telephony boards such as CAHTA, CallURL and Synway cards.

Edit: On second thought, it would be a lot simpler if you just selected the "Send Ext No." option in your PBX settings, instead of send "#6 + Ext No." This way there is no need for an advanced dial string or anything of the sort. It would be a simple case of setting IVM to receive a maximum of 3 digits and store it to a variable like %mailno% and then sending the caller to Voice Mail box %mailno% Using the #6 + Ext No. method: the dial string would be XX111, meaning ignore the #6 and just store the extension number. Then just make sure you have voice mail boxes with the same voice mail box name as the extension number it belongs to. Also make sure you tick "Allow '#' key in data entry" otherwise IVM will take the first digit (the # from your PBX) as a signal that the end of the data entry has been reached. Set maximum digits to 5. Let me know how this works out.

Basically it works like so: Your analogue PBX will send the call to IVM's voice mail extension. Assuming your PBX is set-up to do this, it will then play a series of digits (lets say for example 5 digits) which correspond to what extension was called and is busy for example. IVM will be programmed in its answering OGM to listen for those 5 digits, and after they are entered, automatically send the caller to the voice mail box with the same box name as the string of digits sent by the PBX to IVM. The reason why you must enter the maximum digits that will be sent is because IVM needs to know when the data entry (i.e. the PBX sending the digits) has finished so it can stop listening and send the caller to the voice mail. Otherwise IVM will just wait there for further data input or until the # key is entered (if your PBX dials a # after the digit sequence, you don't have to worry about max digits as IVM will know right away that the end of the data entry has been reached).

Ignore post, see above.

The number of lines in Express Talk is just an indication of the maximum number of calls that particular softphone can answer and have active calls with at any given time. It does not affect how many calls can be received and sent to your voice mail system in IVM (this is completely separate). IVM can answer 64 calls at any given moment. Just three things to point out: 1) IVM should be set-up with an extension in Axon, not set-up with Express Talk. In Axon you can set-up Voice Mail in your softphone's extension settings, and under the voice mail settings you can select the IVM extension to answer the call (when there is no answer on your actual phones). 2) Your VoIP provider will limit the number of calls that can come in at a time. Just because IVM can handle 64 calls at a time and Axon isn't theoretically limited by any number of calls doesn't mean you can accept that number of calls. i.e. CallCentric will limit the number of simultaneous incoming calls to most likely 2 channels (simultaneous calls) at a time. I believe CallCentric allows you to purchase additional channels (kind of like virtual lines to accept more calls) for a fee, but I think it is almost or as expensive as paying for a whole new phone number with them.. so it is up to you. 3) The speed of (or bandwidth available on) your internet connection also plays a big factor in how many simultaneous calls you can send out (or receive) at a given moment. For example on a high quality codec, your highspeed Internet connection may only be able to support 4 calls at a time. So if you plan on several incoming calls at a time, it may be a good idea to invest in a good quality higher speed/bandwidth connection if possible.

That's very strange. I suspect the support person just doesn't realize that they also support GSM. I know that sometimes I find the same thing where they don't list GSM as one of the available codecs in their FAQ, yet it seems it can be used. I'm not sure if it's because it is not used that often compared to the other codecs or why they choose to exclude it. You should be able to turn on SIP tracing in IVM and look in the logs to confirm for sure what codec is being used.

Express Talk can support up to 6 lines at a time, this means that you could have for example 3 incoming calls on your Call Centric phone line at the same time and Express Talk would show and indicate the 3 incoming calls on line 1, 2 and 3 (you can think of them as virtual lines). At the same time, you could then phone out on another available 'line' (e.g. by clicking the line 4 button) and even have a conversation with another phone connected to your Axon PBX on another 'line' (e.g. Line 5). All you have to do is click the line you want to talk on (if it is not in use, you'll get a dial tone for dialing out) and the rest of the active lines will be put on hold until you click back to them. Note that CallCentric does limit the number of simultaneous incoming/outgoing calls you can have going at the same time, but calls between your Axon extensions are not limited. So for example you could have a softphone on your computer, and SIP phone in another room, and your SPA942 in another room again and they could all call each other internally at no cost as if they were on separate lines, yet could all still call out and receive calls from the Call Centric line. Also, another neat feature is that if for example an incoming call comes in and you answer it on the Softphone and then another incoming call comes in after that, not only will the Softphone indicate there is another incoming call, but your other VoIP phones/devices will also ring and can be picked up to answer the second call. Also, someone else could make a call on another VoIP phone on your Axon PBX system without interfering with your or someone else's call.

Well any phone that uses a GSM sim card (as opposed to a 3G or CDMA or other service) will be using the GSM codec. That is still a high portion of phones and GSM isn't being phased out anytime soon. I believe it is the most popular cellular network codec used worldwide, at least for now until 3G networks replace it. In the United States Cingular, AT&T, and T-Mobile all use GSM while Sprint, Verizon, and Virgin use CDMA.

Typically the PBX setup is as follows: You create an external line in Axon for each SIP VoIP account you want to use for incoming/outgoing calls on your phone network. You then create an individual extension in Axon for each softphone (such as Express Talk) or SIP phone/hardware (such as the SPA942). These extensions act as new SIP accounts that can share the same original VoIP account from Call Centric. This is done through Axon which acts almost like your own private VoIP provider, you can think of each extension on Axon as a new Call Centric account, when in fact they are all sharing and have access to only 1 account. The next step is to setup a Dialing Plan that will suit your needs, for example you can create a new Dialing Plan with a rule that says any call that starts with the number "9" should be called on the Call Centric phone line, but without actually dialing the 9. This is the kind of set-up they have in most offices/hotels where you dial a number to get the outside line and you can call internal extensions by simply dialing the extension number. You could also do the reverse for example where all calls are routed through to call centric unless they start with 9, and in that case call internal extensions instead. Also for the SPA942, if it's anything like Linksys' other VoIP products (which I may add aren't exactly user friendly), form fields labeled 'proxy' are equivalent to the server address (internal IP address of your Axon computer), SIP number is the Display Name / Caller-ID, Username is the extension name in Axon, password is the password for the extension. If you need further clarification or instructions, just let me know.

It has been great having you here. I'm sad to see you're leaving and hope you enjoy your time in Europe, USA and especially Canada (I happen to be Canadian ). So is this the final farewell or will you ever be returning? Either way I do wish all the best for you and hope you still keep in touch around the forum.

Just remember that a 100 mbps LAN connection is simply a measure of the bandwidth available in your internal network. The bandwidth of your internet connection is the real determining factor for calls to and from outside callers. If you are unsure of how much bandwidth you have available of your internet connection, you can use a 'speed meter' such as the one on http://speedtest.net to get a fairly accurate assessment. It is very unusual to find an internet connection that can provide 100 mbps in bandwidth, most range anywhere from 256 kbps to 8 mbps. Also I double checked and GSM is the same codec used for cellular telephony, so the call quality will be almost exactly like that of cellular calls. I should warn you though, from my own personal experience, if a cellular phone calls a VoIP phone which is also using the GSM codec, the quality gets pretty bad.