pythonpoole

You are correct that Callcentric has a problem with initiating call transfers (at least with NCH software). I'm not sure of the exact reason, although NCH says it's due to a bug in Callcentric's system. This is the first time I'm announcing this to anyone, but I am in the process right now of starting up my own affordable VoIP service. I plan on letting people from these forums have a chance to test the system out before anyone else does to make sure it's fully compatible with all of NCH's software (which I want to make a priority). My aim is to have: - Full compatibility with call transfers (blind & confirmed) for NCH software - Ability to pick your outbound Caller ID number (since Axon lacks a feature to set one) Other planned services/features: - Affordable outgoing call prices for business and residential users (e.g. 2c/min to anywhere in the USA or Canada, great international rates) - Incoming local numbers with unlimited channels (simultaneous calls) for as low as $4 a month plus usage (numbers available for all over the world) - Toll Free numbers available for many countries around the world (including USA, Canada, UK & Australia) starting at just $4 a month plus 4c/min incoming - e911 support for emergency services (USA) - Online control panel to manage your VoIP account - Support for: G.711 ulaw/alaw, G.726, G.723, G.729a, iLBC and gsm codecs - Standard extra call features (including voice mail, call waiting, etc.) - Unlimited outbound channels Let me know if you're interested.

Not yet, at least not that I'm aware of (I haven't yet tried out the latest version)

This is the upgrade page here: http://www.nch.com.au/upgrade/index.html After entering the correct registration details, it should provide you with upgrade pricing. Note that if you do not enter correct registration details, normal (full) prices will be shown.

If you are using IVM you can send the call to IVM's voicemail extension directly. Also some Softphones have a DND (do not disturb) function which I think sends back an 'unavailable' message to trigger voicemail. If it doesn't go straight to voicemail, at least it won't disturb the person with the softphone while it 'rings'

I don't think Axon can do what you're asking. It can only ring extensions, not outbound numbers... However, I think I've come up with a plausible work around, although definitely not the best solution: 1) Create an extension for each outbound number you want to ring upon an incoming call 2) Create a ring group with all the extensions added to the list 3) Set-up multiple SIP/VoIP soft phones (which have an auto transfer/forward feature) with the extensions you created in Axon, and then have them set to transfer to the appropriate phone numbers. Now this is where you'll have to test the set-up because it's based on theoretical hypothesis (so to speak), so there is no telling how it might actually work out in a real world environment. What I'm concerned about is that Axon may be unable to signal the end of a call (e.g. if the caller hanged up, or if the call was taken on another extension/phone number); but I can't tell from planning whether or not this will occur. I suppose it will depend on the soft-phone being used, if it disconnects a transferred call if the original call stops ringing, it should be fine.. but if it just forwards the call blindly, the phones might keep ringing even if the call has long been answered or disconnected.

Seems strange to me. I know IVM used to have a limit on the amount of numbers which could be in the list, but I seem to remember that limit was removed in the last 2 or so versions of the software, or at least the most current one. What version are you using?

Well to begin with, NCH offers its own conferencing software called Quorum. While it doesn't give you as much control over the conference as some other conferencing apps do, it does provide some functionality that is rarely seen in other apps. For example, users can join the conference from a VoIP phone / soft phone, a land-line (assuming you have set-up a VoIP inbound DID number directed to Quorum, and last but not least it's possible to join the conference from any computer with a web browser, headphones, and microphone using an online web interface (and I believe an embedded Java applet). Quorum also provides an easy to set-up interface (both through the application itself, and through the web interface. It's also easy to set-up with NCH's other software like Express Talk. Another conferencing app to look into is the Conference module that is part of FreePBX (which runs off the Asterisk PBX). This conferencing app is pretty basic, but it does provide some of the smaller (and useful) features that Quorum doesn't. For example, you can have it announce or play a ping sound when users join and leave the conference. You can also provide users an admins with a menu they can access during the conference that lets them access features such as muting, adjusting volume, blocking users (admins/conference leaders only), and so on. The conferencing app of FreePBX also has the advantage of being free, which Quorum is not, however it lacks a web interface for people to join the conference on, which seems to be important for your case as you said 'web conference appliance.' Also quorum can run of an existing Windows computer just like any other program, where as Asterisk (with FreePBX) requires a dedicated server (for the most part) running Linux.

Which version of IVM are you using? I believe the older versions of IVM had a limitation, but the most recent 2 or so versions shouldn't have the problem. Also I don't believe the limit was ever that low though, so it could be another issue.

Wendy, you can't just use and old data/fax modem with IVM. Because IVM is used for voice telephony applications and not just data streams, there are some specific requirements for the hardware. These requirements are: 1) The modem is TAPI compliant 2) The modem has a Voice feature, often labelled as "voice modem" They tend to sell for around $30 (compared to a regular data/fax modem which sells for $5 - $15) Any modem which does not meet those requirements either won't show up in IVM's device list, or will simply fail to work properly with the software. Please check with the manufacturer to make sure these features are supported before purchasing the modem. Also, this page on NCH's website will help you determine the compatibility your modem (or one you plan to buy): http://nch.com.au/ivm/modems.html

Wendy, you can't just use and old data/fax modem with IVM. Because IVM is used for voice telephony applications and not just data streams, there are some specific requirements for the hardware. These requirements are: 1) The modem is TAPI compliant 2) The modem has a Voice feature, often labelled as "voice modem" They tend to sell for around $30 (compared to a regular data/fax modem which sells for $5 - $15) Any modem which does not meet those requirements either won't show up in IVM's device list, or will simply fail to work properly with the software. Please check with the manufacturer to make sure these features are supported before purchasing the modem. Also, this page on NCH's website will help you determine the compatibility your modem (or one you plan to buy): http://nch.com.au/ivm/modems.html

Yes, in fact using VoIP services (such as MagicJack) with IVM and it's autodialer feature is common. Usually however the VoIP service uses an open protocol (e.g. SIP) which allows IVM to communicate with it directly, elminating the need for a voice modem connected to the magicjack which is then connected back to the computer (unnessecary redundancy anyone?) Granted, most VoIP services aren't completely 'free' like MagicJack but there are many who charge very little for calls (e.g. unlimited north america for $20/mo, or 1c/min to certain destinations). Without proper configuration, most VoIP services will provide a similar quality to what you described for Vonage. This is not unique to Vonage, and if you configure your network devices properly and tweak Vonage's settings you should have no problems with audio quality, nor bandwidth issues (slow downs). For future reference, these are a few tips for improving audio quality: - Configure your internet router to forward ports used by the VoIP device - Enable and configure the 'Quality of Service' feature on your router (if available) - Go into the Vonage online control panel, and change the bandwidth settings to use low bandwidth - If you are on a low-end high-speed connection, consider upgrading to a 'faster'/greater bandwidth plan

I don't know if Axon supports this, but a , comma can be used in some of NCH's other software (e.g. IVM) to signify a pause of about 2 seconds. In the dial string standards I think W can be used to wait for a dialtone, but I doubt Axon would support that feature even if it did recognize the comma.

I believe the Video to Video feature is dependent on your VoIP provider. If both the provider you and your friend are using support video calls, then it should work fine. If not, it probably won't work. What provider are you using?

You can also set Axon to prepend 1 or more digits to outgoing calls through the dial plan set-up. This could allow you to automatically dial the password before sending out each call.

Under 'Options' -> 'Network' tab -> 'Use static IP address and static mapped ports (Advanced Option)' You can then set the IP address and port numbers manually. If you select this option, you should deselect 'Use PNP...' and deselect 'Use STUN...' as they become unecessary and may still cause conflicts when resolving a public IP address.

No, but they did offer me a product of my choice from their website once, I accepted. You can report software bugs directly here: http://www.nch.com.au/software/bug.htmlYou can also contact NCH about Axon here (bottom of page): http://nch.com.au/pbx/support.html I remember once having a problem where the extension would continuously ring after the caller had disconnected. Unfortunately I can't remember how I solved the problem. Also, it only happened with ET I think, and now I use hardware based phones and/or VoIP ATA adaptors which perform well and have greater stability (in my opinion). I would say you're correct in suggesting that Axon should interface directly with IMS and use that for MOH when an extension initiates an MOH request. Unfortunately it doesn't work like that, at least not in this version and I believe the client phone (i.e. ET) is supposed to hook-up directly with IMS with the option you mentioned.Note, you can download Express Talk business edition free to try for 14 days (last time I checked). If you already installed ET or ET business on the computer, use another computer for testing. You should be able to use the IMS on hold feature to test it before purchasing the product. If you can, try installing Axon on another test computer with similar specifications (and if possible, running the same OS), monitor the CPU usage and see if it reports similar results. Perhaps there is some other software running on the OS that is causing a conflict, I'm not really sure what to make of this, in fact I'm not really sure what to make from a lot of what you have said (most of the problems are new to me). What I will say is that these products were intended for (and I believe are also suitable for) small to medium sized businesses and home offices. For large corporations or high reliability services (e.g. Emergency Call Centres, Help lines, Hospitals, etc.) if I did decide to use NCH products, I would make sure I had back-up/redundant systems which could be deployed instantly in case of problems. Your welcome.

If that's for real, that is hilarious. I actually laughed out loud . The basic way of configuring any SIP (FXS) ATA adaptor is: 1 - Create an extension in Axon and assign it a numeric username (Extension ID) and password 2 - Use the VoIP ata's web configuration to set-up an SIP VoIP service to connect to - - Use the username and password you selected earlier and specify the Axon computer's IP address for the domain/server/proxy 3 - Create an external line in Axon that will connect with your VoIP provider and fill in the relevant username, password, and extension you want incoming calls to ring to. If you are trying to use the ATA as an FXO adaptor instead, then the instructions above do not apply. Connecting an FXO adaptor to Axon is heavily dependent on the equipment and I suggest referring to the product manual. If this is what you're having trouble with, you can post again with further details but I'm not sure I can be of much help as I've never worked with a Zoom ATA, especially not as an FXO. *An FXS adaptor is used to receive and make incoming and outgoing calls respectively through a SIP VoIP service or PBX (like Axon) - This is how I assumed you wanted to use the adaptor *An FXO adaptor is used to connect a PBX (like Axon) to normal PSTN analogue phone lines (for incoming/outgoing calls on the analogue line)

The first problem you have is that Axon is having trouble auto-detecting the Public IP. This could be due to ports being blocked on your broadband router (or firewall). Please ensure that you forward the port(s) used by Axon (see your router manual for more information on how to forward ports). Although not always required, port forwarding can improve performance significantly and can solve many problems, especially those related to Audio transmission. Ok, the next issue is the "404 Not Found" error. This error occurs when the number you are dialling does not exist, or more likely, if the number was dialled incorrectly (e.g. missing country code, area code, or outbound dialling code). Please make sure your outbound routes are set-up correctly to dial on the VoIPtalk, and also ensure that you are dialling the number as per the instructions from VoIPtalk. Lastly (probably obvious), check your account balance to make sure you can complete the call (although you'd probably get another error if this was the problem).

Yep, looks like it should work fine. Almost any SIP compatible VoIP device should work fine with Axon, IVM, etc.

What SIP Voip Service are you using? Do you know if it supports call transfers? Check here for more info on whether call transfers are supported with your VoIP provider

Sorry, you sent it through the forums and I tend to ignore those e-mails as they're usually just to let me know someone has posted a new topic or something, in which case I just check the forum for new posts. Replying to you now.

I wonder if it is a manufacturing fault with the USB phones themselves. For me, mine work as expected... but I can't seem to get the speaker function to work at all with Express Talk. A bit strange. I suppose it could also be Express Talk since there was a period of time / version of ET where my USB phones wouldn't ring, and then all of a sudden when I upgraded, they started ringing as expected.

Express Talk supports G.711 (alaw/ulaw), GSM, and I think it also supports G.726. It's the same for all of NCH's other VoIP software. The reason NCH doesn't support G.729 or G.723 is because of their proprietary nature which would require the purchase of a license for each person using the software (making it impossible to provide for free, and also increasing the price of the business edition). I have no problem with the quality of Express Talk calls, but here are some tips to help you out: G.711 provides crystal clear audio quality, but will break up the voice if there isn't enough bandwidth available (be warned, it uses a lot of bandwidth!) G.726 provides good quality clear audio, using much less bandwidth (with less or no voice break-ups) GSM (same codec used on GSM cellphones) provides satisfactory audio quality (not as clear) using very little bandwidth (virtually no voice break-ups) The quality of audio can also be affected by many other factors. To improve audio quality and reduce voice break-ups: - Be sure to forward the necessary ports on your router/firewall - Be sure to have a stable Internet connection that is relatively fast (consider upgrading if you have lower-end hispeed) - If your router supports it, be sure to enable and set-up the Quality of Service feature. This will help put priority on VoIP traffic over regular Internet traffic Other factors that may affect the quality (things you can't really change): - The distance between you and the VoIP server (the larger the distance, the larger the delay, and also the greater chance of audio quality degrading) - How your ISP decides to manage your Internet traffic. If your ISP doesn't put priority on VoIP traffic, it could result in broken audio - The bandwidth available on the server end (of your VoIP provider) could be causing voice quality - The codecs allowed by your VoIP service could be limiting your choices to help save bandwidth or improve audio clarity

An SIP account is used to connect a client's VoIP phone (or software like Axon) to a VoIP service. SIP itself is a protocol or set of rules used to communicate between voip devices. The account is just a series of login credentials (e.g. username/password) to authorize that communication. SIP accounts are configured with Axon as 'External Lines' and can be set to ring to a particular extension or group of extensions on incoming calls. Axon can also be set to use an SIP account / External line to make outbound calls. In most cases an SIP account is linked to a single phone number. By having 1 SIP account for each phone number, Axon can be set to handle and route the calls differently for each line.

Is there any way you can have the two phone numbers put on separate SIP accounts with your VoIP provider? As it stands now, Axon won't be able to tell what number the caller phoned in on, and thus won't be able to decide whether to route the call or ignore it. It is very dependent on your VoIP provider, but most of the providers I have used allow you to set-up multiple SIP accounts under the same general 'account' on their system. Then each phone number can be linked to an individual SIP account, and Axon can deal with each phone number separately.