pythonpoole

Well I can't see how it would be possible to guarantee you exactly 256 kbps. Unless they feed 512kbps in your home and have programmed your router to limit your connection to 256kbps or something. DSL connections drastically reduce their bandwidth with distance. I have an 8,000kbps connection, and I only receive about 3,500kbps at any given moment in bandwidth because I am 2.5 kilometers from my telephone exchange. There is a rule of thumb that after 4 kilometers the DSL connection quality will be so bad that you will be barely able to use it. Now if you lived within a couple of hundred metres from the exchange, I would expect you would have pretty close to 256kbps in bandwidth. Likewise if you were using a non-telephone/dsl connection the distance may pay less of a factor. (E.g. if you have a Fibre Optic internet connection, you would most likely be able to use most of the bandwidth provided to you. I suggest you use a speed meter on your end (e.g. the one on http://speedtest.net) as this will give you a relatively accurate measurement of the actual bandwidth you have available to use.

Great news I just have a couple of questions? Does the video feature work when using Axon. (e.g. can one extension (using Talk) video call another Axon extension (also using Talk)? Or is it just Talk to Talk?) What sort of compatibility is expected in terms of video calling other non-Talk VoIP users (i.e. users with other video enabled VoIP hardware/software)

- Echo Having an echo is normal and can be fixed. You will usually get an echo if you are answering the phone with an open microphone and speakers. You should reduce the echo to nothing if you instead wear a headset. The echo is caused by your voice entering the microphone, then coming back out through the speakers and then entering the microphone again. If you are still experiencing an echo problem, make sure you select the feature "Use microphone and speaker and turn echo cancellation on" in Express Talk's audio options. - Delays Delays are also normal, but are usually very small. An average delay for VoIP communications is probably around 0.3 - 0.5 seconds (depends on a lot of factors). Large delays can be caused by: - Large distance from you to VoIP provider (i.e. if your VoIP provider's SIP server is in another country, you will most definitely experience much bigger delays than using a local VoIP provider that runs from your city. Remember that the larger the distance, the larger the delay. - Slow internet connection / limited bandwidth. Make sure you have a good quality internet connection with high speeds / large bandwidth. E.g. I would never recommend a 256kbps connection for people who wish to use VoIP, that is the absolute minimum. I recommend having a connection rated at around greater than 3mbit. The faster and greater bandwidth available, the smaller the delay and the less amount of quality issues in terms of cutting in and out. - Having a modem or router that does not support QoS. QoS is Quality of Service and is a feature on some higher quality modems and routers that enables you to prioritize VoIP communications data over other network data to help reduce quality loss and also ensure VoIP call data is sent ahead of other less important data (e.g. regular web browsing), If you are not using a router with QoS on, I suggest you enable it, or buy a router which supports it. - Poor Quality Like delays, poor quality can also be caused by many factors. In fact, many of the same factors for delays are also factors which affect quality loss, such as: - Internet connection quality / speed / bandwidth available - Whether QoS is enabled on the routers in your network - Distance from you and your ISP, distance to the VoIP provider One other thing that might help is Express Talk's "Prefer lower bandwidth usage (But with lower audio quality) Although the actual maximum quality of the audio is reduced when this setting is selected, much less bandwidth is used. This means that the call should not experience cut outs, or connection problems where you can hear the person one second and not the next. This may be a viable option in your situation, but essentially it comes down to whether you want pure clarity and perfect quality but with voices cutting in and out, or reduced clarity, a small bit of fuzziness, but a consistent connection that doesn't keep dropping in and out.

For many reasons: - Telemarketing calls - To quickly inform people of something (e.g. you could get IVM to immediately call several numbers and deliver an emergency message) - You can have IVM make automated phone calls to people on request from other applications (e.g. if you are in a library/video store, you can have your library/video store management program automatically tell IVM to phone people with overdue books/videos) - You could even use it for Wake-up calls There are many other uses for the feature, those are just some examples.

Yes, this is possible. In Axon, create an extension (or use one you already have and edit it) that incoming calls will ring to. In the Voice Mail category, make sure you un-tick Use voice mail if not answered or busy In the Transfer if Not Answered category, tick Transfer the call if not answered, and in the Transfer to Number: textbox, enter the extension that is set to go to your IVM interactive menu. (If you don't have an extension for this yet, simply enter an unused extension number and then create an extension after for your IVM menu later). Then select a Time before transfer (seconds). The default (14 sec) should be fine.. also it may be important to keep it under 20 sec (explain why below). If you want to have several extensions to ring when an incoming call comes in you will need to create a Group/Queue and then add the extension numbers you want to ring (make sure you include the extension you created above). Note: If you do use multiple extensions and use the ring group method, you should make sure that either Voice Mail is turned off for all extensions in the group OR that the Time before transfer (seconds) is less than the time that Voice Mail kicks in for the other extensions (default 20 sec), otherwise callers may be directed to the voice mail system rather than the interactive menu.

Express Talk (and all of NCH's other VoIP products) support two main protocols: G711 and GSM G711 provides very good quality, but it uses a lot of bandwidth (~64 kbps or more) GSM provides good quality and uses much less bandwidth (~13 kbps) The thing is you will never get the full bandwidth you are allocated, especially when you're talking about an external ISP. For instance, I have an 8.0 mbit internet connection, but on average I can only use up about 4 mbit of bandwidth at a time due to external factors such as distance, quality of the connection, the amount of devices accessing the network etc. So if you have a 256 kbps connection, even if your ISP was of a superb quality, and even if you lived next door to the telephone exchange where your internet connection is routed through.. you shouldn't expect to ever get your full 256 kbps worth, and should probably assume at any given time there is only about 128 kbps to work with. However, if your modem/router supports QoS (quality of service) you can help prioritize VoIP call data so that it is guaranteed to get as much bandwidth as possible by limiting the bandwidth of other internet data (such as checking e-mail, or normal web browsing). So what I'm saying is don't push it, you probably have a lot less bandwidth than you think. Tip: You can using speed meters available on the net (e.g. speedtest.net and the one at 2wire.com to estimate your actual available bandwidth). Let me also add this: Most LANs have a bandwidth rate of about 100mbit, so you should have no problems handling several calls between users on your LAN. The thing to be worried about is several simultaneous calls to people outside of your local phone network. Remember that the bandwidth is only used when there is an active call (only very minimal data is sent when there is no call in progress). So in order to use up all your bandwidth you would have to have several calls taking place during the same time period.

True, but it's not really what the topic poster is looking for. In some situations (like this case) the director will have their own phone number, and they will answer almost all of the calls that call in on that number.. but very occasionally his assistant might need to take one of his calls, even though the assistant's extension doesn't ring for calls on that line (due to the volume of calls and the fact that the assistant answers only very occasionally). So this is where a feature to pick-up calls that are ringing on one extension from another non-ringing extension is useful. This feature is common to most other PBX systems including Asterisk, and is supported by many IP phones (e.g. the entire Linksys/Cisco range). Unfortunately Axon doesn't support it as of now, and the only way of answering a call from another extension is to also have that extension on the same ring group. What you can do is set a delay on the assistant's ring, so if the director is not in his office, the assistant's phone will start ringing after a predetermined amount of time (e.g. 15 sec), thus minimizing the disturbance to the assistant during normal day-to-day phone calls that the director answers.

I am a bit interested in trying out a billing solution for Axon. However If I was to purchase in the future, it would most likely be for a very small scale system. By the way, the link you gave for the form appears to be broken (standard 404).

The SIP messages sent from your computer's softphone to the VoIPwise server also contain information about what type of software is being used on the client (i.e. your) end, meaning it would not be difficult for them to set the system up so the free calls offer only applies to their softphone. There is no mistaking that they are intentionally trying to do this, and you probably won't be able to get the free calls offer with any other software even if you do ask their support team. The only way in which I can see "beating the system" if you will, is to modify Express Talk so that it sends out SIP messages under the software name of the VoIPwise softphone. Firstly there are many problems with this: 1) I am quite sure that NCH is not going to do this for you (for various reasons, including possible legal implications) 2) There is no guarantee it would work, this is only one of the ways in which VoIPwise could be using to determine if the user is using their softphone (another way they may be able to do this is via additional TC/IP messages sent between the client and server that are separate from the SIP communications). 3) Any attempt to modify the software yourself (e.g. using decompilers, resource editors, hex editors, etc.) in anyway will surely go against NCH's terms of conditions / use agreement.

Many of Yealink's products have similar drivers, and you may find that some functions of the phone will operate fine with Express Talk while other features may not. You'll have to try it out for yourself to see though. Technically, the P4K model is the only one officially supported by the software, although even that model has problems ringing (at least for me (on XP), and also NCH noted the problem when running Vista).

I am having the same trouble. What is annoying is that if you try the same thing with an internal extension, you do hear the ringing and/or hold music set-up in Axon.. but for some reason all incoming calls from an external line just go "dead" (but still active) during a transfer... I have tried a few things and still haven't been able to correct the problem unfortunately.

NEYLORUSA, what is the problem with it? All the log shows is that the call was transferred, and rang several times before being disconnected (i.e. no one answered). Is it not transferring at all?

The carousel FXO adapter is only compatible with professional Telephony boards (such as those listed on NCH's website). The voice-modem option is very limited in its functionality in that it is only good for recording audio (e.g. voice mail message) and maybe an automated outbound call, but that's about it. It isn't designed to interface with PBX call management solutions or anything similar.

The executables fetch.exe, confirm.exe, and cancel.exe can be found in the directory C:\Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM

There are several good VoIP providers out there, however many of them don't handle simultaneous calls, or they charge extra for that privillege. Here is a list of recommended VoIP providers from NCH: http://www.nch.com.au/talk/sip.html From my experience, Call Centric has poor quality and large delays, however I am in Australia so distance does play a factor (although I don't have the same problem with my other VoIP lines). I've also heard that inPhonex is touch-and-go with accepting key digit presses, so if you plan to use it with an IVR, try and see if you can get a free / demo account to try out.

I thought I should let you know, that unfortunately, I just discovered that FreeDigits doesn't seem to be offering the same free service to new customers. I still have a free account with my own number, but now they changed the name of the site to "Ring-To Number" and it says the free version simply gives you a shared number where the person inputs a code/extension number to forward the call to you, where as the paid version gives you a private number (which used to be the free version). FaxDigits is also part of the same company. What you can try to do is sign-up with them, and then in the control panel click the menu options to add a phone number. (This is how I did it a long time ago) and it would automatically add the number through FreeeDigits. So you can try to do that, and see if it still lets you, otherwise you may have to pay the "premium" rate of $5/month. In which case there are other services around which may be equivalent or better for around the same price.

Unfortunately, if you only have one Vonage line and it becomes busy, Vonage will automatically forward the call to your Vonage voice mail. However, you can change settings in Vonage's online control panel to for example forward calls to another number in the event there is no answer.. Also, there is a very good and free service called FreeDigits ( http://freedigits.com ) that will allow you to have a private US Phone Number for free, and it also offers concurrent calls. This means that you can set-up Vonage to forward incoming calls to the FreeDigits number, and then have your FreeDigits number connected to Axon through the VoIP account they give you. This way you can have for example: two different callers call in at the same time as well as someone else in your business using the Vonage phone line to dial out. Keep in mind though you will only be able to use your 1 Vonage line for dialing out, this method simply allows you to receive more than 1 call at a time, and make a call while callers are still on the line. This is exactly what I have set-up at home, and It works pretty well. Although sometimes the Caller-ID appears as 'FreeDigits'. The service is reliable, but of course it is up to you if you want to rely on a free service to manage your inbound business calls. If the budget is available, it may make more sense to either purchase additional Vonage lines, or go with another VoIP provider that has an 'open' system whereby they give you the VoIP account information for you to put into Axon. Vonage is mean in that sense.. they lock all of the devices they give you and don't let you view your own account information.. for the sole purpose of stopping you from using and purchasing other non-Vonage hardware to use with the account. There are only a couple VoIP providers that do this, and I don't understand why Vonage has to be one of them, but what are you going to do about it. Also I was wondering how on earth you managed to get an unlimited line for $15 ... I use Vonage Canada, and they charge $40 a month for the unlimited (residential) plan! Keep in mind the Canadian dollar is now worth a tad bit more than the US dollar.

So, if I need 3 extensions do I need 3 separate phone lines? Using Axon from NCH (free software), you can add and manage any number of extensions for a single or multiple phonelines. Each extension can be used by either a computer softphone (with a headset or external USB phone), an SIP phone (connected directly to your internet router or switch using Ethernet) or a traditional analog phone using an ATA adapter connected to your Internet router or switch. Will my Vonage phone line work or do I need to signup for Vonage Softphone? In order to connect your current vonage line to NCH software, you will need to purchase an FXO adapter. This converts any analog line into a VoIP line. Even though Vonage is a digital VoIP service, the output from the ATA device they provide you is Analog. Essentially this means the call will be converted from VoIP to analog to VoIP again. Fear not, because I am using this exact method and it does not cause any noticeable delay or quality loss. The (Linksys) Sipura 3102 FXO adapter is around $80-$100 depending on where you purchase it. You can also use the Vonage soft-line and connect it directly to IVM's software, however Vonage soft-lines don't have unlimited plans, and you can only get them in combination with their regular service. So, the soft-line is essentially a rip-off and financially doesn't make sense unless you absolutely need to use the features of a soft-line. Can I use Skype in combination with Vonage for extensions? Skype can interface with IVM software (using Uplink) in such a way that incoming calls from Skype can be routed through to call your internal extensions. Additionally, you will be able to make outbound calls through your Skype line, and you can also forward calls to another Skype number. However, that is pretty much the extent of which Skype can interface with this software, you won't be able to simply add Skype extensions to be rung like your other extensions on incoming calls for example. Also, I warn you that the Skype interface is not completely 'bug-free' in my opinion, and still needs work. Judging from experience and the topics in the uplink forum, there seems to be common problems such as disconnections when there are multiple concurrent Skype calls, and an inability to detect key presses over a Skype line. Will I need Axon or can I just use IVM independently? You can use IVM independently, however the software basically operates as just an answering machine / basic IVR on it's own. If you want other features like multiple extensions, advanced call routing, dial-plans, call forwarding, dialing groups, etc. you will want to use Axon in combination with IVM. There is also additional 'plug-in' type software for extended phone features such as Music-on-hold and a conference centre. -- Edit -- Other useful information: - Each extension can dial another extension in your phone network as well as use external phone lines. - You can set-up dialing plans and control exactly how each extension (or a group of extensions) can make phone calls. For example, you can specify that Extension 1 can make both internal or external phone calls and Extension 2 is limited to only external phone calls. You could also for example prevent certain extensions from making certain types of phone calls, such as international, premium, or mobile phones (depending on your country). - Extension groups can be set-up to allow multiple extensions to be considered one extension when an event occurs. For example, if I setup a group with extension 1 and 2 in it, I can tell Axon to ring that group of extensions when an incoming call comes in.. otherwise only 1 extension could be rung. Additionally, there are more advanced dial groups which can have for example different delay times for ringing. For example, ring one extension for a specified number of seconds, and if no one answers automatically start ringing another extension. - There is virtually no limit* to how many external lines you can have.. meaning if you wish to upgrade your system in the future and add additional VoIP lines (or FXO lines), this is quite possible and easy to do. You can use both lines together, and for example even set-up a system where if one line is busy, the call can automatically use the next line that is available. Don't hesitate to ask if you have any further questions.

EDIT: I forgot to add that an FXO adapter will not disrupt the functionality of the other (traditional/analog) phones on the line. Unfortunately there are several problems with using modems for telephone communication and call management systems (whether or not you choose to use NCH products won't make a difference): 1) The modem would have to be a voice-enabled modem.. the standard data/fax modems found in most computers do not have the capability of handling phone/voice applications such as answering machines. Voice-modems are also more expensive than traditional data or data/fax modems. 2) Voice-Modems typically provide poor and inferior quality compared to professional voice/telephony boards and FXO adapters, and also lack many basic features. 3) Even if the modem in question is voice-enabled, this only lets it connect to IVM (or other telephony software) through a direct hardware connection (e.g. for an answering machine), meaning most other features such as allowing remote computers to make calls through the phone line, having software to manage the flow of phone calls and route them through to different extensions would not be possible. The only way to achieve the functionality you were asking for in your first post (in terms of remote computers operating the phone line, and receiving caller-id etc. etc.), is to either use a professional voice/telephony board (which is essentially just a much more advanced version of a voice-modem) or an FXO adapter (my personal preference). The voice/telephony boards are also typically more expensive (Range from about $250-$800) FXO adapters are much cheaper, but unlike most voice boards can usually only handle 1 line per device ($80-100).

Instead of using the modem options, I suggest using an FXO adapter such as the Linksys (Sipura) 3102 FXO adapter. This product is essentially an adapter that converts a traditional landline into a VoIP phone line. This makes it much easier to share a phone line across a computer network and setup a "virtual phone system." As for the software, once the line is VoIP (converted with the FXO adapter), IVM's Axon software can manage the call easily. Axon (Free) lets you manage your phone system easily, by providing an easy to use interface that enables you to direct and route calls between different phones and softphones (computer software phones) connected to your system. It is essentially the heart of the system. IVM Answering Attendant acts as your answering machine and attendant software (e.g. Press 1 to do this, 2 to do that). It is very simple to set-up and has one of the easiest interfaces for programming answering attendant menus and data collection etc. The nice thing is NCH's software all sort of just 'clicks' together, so you can easily setup Axon to forward calls to your IVM voicemail system after the call hasn't been answered for a period of time. Express Talk (available in Free and Business Edition) enables users to make (and take) phone calls from your landline (assuming you have an FXO adapter) from any computer on your network. The software can be used with a headset, or is even compatible with external phones that plug-in to the computer's USB port. The Express Talk option will automatically pop-up the Caller ID for incoming calls. The other nice thing about using Axon and several computers with Express Talk is that each user with Express Talk installed can have their own virtual extension / phone line. There is absolutely no extra cost for the number of extensions you have and there is no limit to how many you can make. This allows each user to have the ability to call any other user on the phone system, as if they had their own phone line, as well as use the outside landline. You can also program a more advanced system with Axon. For example you could incoming calls on the landline automatically ring only the reception phones, but if nobody answers on the reception phones for a predetermined amount of time (e.g. 10 seconds), have the system automatically start ringing another office which can take the call instead. Pretty much whatever you want to accomplish with your phone system can be achieved with IVM's software.. whether it is a simple answering machine, or a complex call center with music-on-hold and interactive assistance menus.

I haven't heard of this particular problem. However, I felt I should comment about how often I keep hearing (a lot!) that putting a call on hold and then taking it off hold solves virtually any audio problems with Express Talk :confused:. I can't determine what the technical explanation is, but it must somehow adjust some audio settings that for whatever reason are not adjusted at other times during the call. I really think the NCH programmers should take a deeper look into this and figure out what exactly about the hold button makes it so magical. I've heard of people who can't talk between Express Talk users at all, but if they put the call on hold and then take it off hold they can speak to each-other just fine.

I don't know if this is in any way related... but every time a network cable is unplugged, or there is an internet connection disruption for a period of time.. it seems to me that IVM crashes each time with that same error. And it is not just IVM, Axon also crashes at the same time with the same message. Occasionally Quorum also fails but other times it does not experience an error when the other two programs do.

Inactive just means the line isn't currently in use. That doesn't indicate a problem unless they remain inactive when someone calls in.

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Some providers have a limit (even if you get the unlimited plans) for how many outbound calls you can make in a given time period. I don't know which ones have limits, but you should contact them before purchasing to ensure there are no problems. Additionally some VoIP providers have separate plans for users intending on making phone calls for business purposes.