pythonpoole

Strange, the confirmed transfer should ask the person being called to confirm they would like to answer (similar to when you receive a collect call from an operator). The answer detect may not work with Swiss ringing tones, I'm not exactly sure, but I can't see why IVM would assume the transfer was confirmed without getting the response from the callee.

Just responded to your PM, check your inbox.

The first thing I would like to point out is that you should follow the ISO (International Standard) for IVR input. In this case, you should use 1 for Yes, and 2 for No. Zero should never be used for input, under the ISO, it is strictly reserved for operator/assistance access. As for your question, it would be a simple case of sending the caller to the QuestionX OGM when they press 1 and the QuestionY OGM when they press 2. On those question OGMs you can then decide to continue to flow out into separate sets of OGMs or have them continue on with the same set of questions. It also depends how you have your survey set-up, are you recording the answers to a database using a plug-in or logging them using log feature in the advanced tab?

To link your VoIP provider with Axon, open the Axon web configuration panel and add a new 'External Line'. Fill in all the details (username/password/server) given to you by your VoIP provider. Once you've saved the configuration, Axon should be ready to make/receive calls. The next thing to do is set-up some extensions. This is pretty straight forward, add a new extension, give it username/number and a password and then you can set-up an IP phone or Soft Phone (e.g. Express Talk) with those details and they should be ready to phone out & accept calls through Axon. The next thing to do is to add all the extensions to a ring 'Group' so that incoming calls will ring all the extensions instead of just one (you can pick and choose which extensions to ring). Once you've done that, simply go back to the external line settings and set the line to send incoming calls to the ring group you just created. [incoming calls should now ring as expected] The last step is to configure a dial plan. Again this is pretty straight-forward, you can basically add rules so that calls starting with a certain digit go out on a certain line. For example, you could set a rule where calls starting with 1 use the internal Axon [Extension] line to call other extensions, and make all other calls go out onto your VoIP line. [Outgoing and Internal calls should now be working] That's it. The only thing is you mentioned PSTN. Generally speaking, PSTN is a term to describe analogue telephony systems and not VoIP. Do you wish to connect normal analogue lines to Axon as well as VoIP lines?

Under Outbound -> Settings... you can configure various answer settings which will adjust the delay time. The delay is there so that IVM can detect things such as whether a human or answering machine answered. By reducing the time taken to detect silences and answering machines, you should be able to reduce the delay before your outbound calls play the OGM message.

If you mean each Telstra line you have is set-up as a different external line in Axon, then you'll need to set-up a dial-plan which allows you to select the outbound line to use when dialling a number. For example, you could have a dial plan where numbers starting with 1 go out on line one, 2 to line two and so on. Then you could select which line to use when you dial outbound calls so that the number corresponds with the extension you are using. Note that all of Linksys/Sipura's products (to my knowledge) support dial plans that let you prepend digits in front of numbers. You could have each Linksys/Sipura phone or adaptor you have automatically dial the corresponding prepend digit prior to making each outbound call so that the correct line is used based on the extension that is calling. On a side note, spoofing is a technique used to mask your true Caller ID with another name/number of your choice. On many VoIP systems this is very easy to do, the client (you) digitally sends the outbound caller ID information to the VoIP service which then sends that on to the analogue phone networks and ultimately to the party you call. On traditional analogue systems there is no way to spoof your Caller ID or send a new Caller ID to the telco. Instead, the telco manages everything for you and has complete control over what Caller ID is used. The only known ways of spoofing analogue networks are: - Dialling into another phone system and gaining access to their dial-tone and making a call from their line - Calling into a VoIP system which can transfer your call to another number using a Caller ID of your choice Also on a side note, spoofing has many legal restrictions and in some regions of the world it is now illegal (even if you also own the number you are spoofing)

I do not believe Telstra permits you to 'spoof' outgoing caller ID. Usually you can only set the outbound caller ID on VoIP lines, but for landlines its controlled entirely by the telco. If you do not wish the other party to see your Telstra number when calling, consider adding the 'private number' feature to your line.

Enabling STUN servers would probably help, they should identify your public IP address and free port numbers to use for the soft phone. You should also be able to enter this information in manually (at least you can in the regular Windows version) if need be.

I'm not sure how you set it up like that. As far as I know, Express Talk just uses the next available virtual line to show the next incoming call. Even if you have 2 SIP providers my understanding is if there is no call in progress and someone phones on the second VoIP line, Express Talk should signal the call on virtual line 1. I will say this, I have not done much testing with Express Talk using several VoIP providers at once, but I know that in my set-up Express uses the 'next available' virtual line approach (I'm using ET with Axon though, so it could be a bit different). I should note that there are other ways to distinguish the call. For example when you use Axon (free for <10 extensions) in conjunction with Express Talk, you can connect an unlimited number of separate VoIP lines and set each one to have their own Caller ID prepend string. This allows you for example to add a 1 in front of line 1 calls, a 2 in front of line 2 calls, etc. When the calls come into Express Talk the Caller ID will be displayed with the Caller ID prepend string to indicate the phone line that was called. Using Axon also has other advantages, for example you can share the phone lines with more than one Express Talk soft-phone or other compatible VoIP phone. Axon also gives you more power over call routing and can do things like play a music on hold file while callers wait for the call to be answered.

Try re-installing the free version. I seem to remember on the older Windows versions of the program there was a 'bug' where the free version would keep expiring after 14 days until reinstalled. I believe this was fixed (although I'm not certain), perhaps this 'bug' still exists on the mac version.

Keep in mind these are "virtual" lines. For example lets say you have 20 physical phone lines. When there is an incoming call on any of those lines, Express Talk will simply use the next available virtual line to signal the call. Express Talk may be limited to accepting only 6 calls at any given moment, but you can have as many physical phone lines as you wish and they do not need to correspond to the line numbers on Express Talk. So unless you plan on having someone answer 7 or more calls at the same time, extra virtual lines shouldn't be necessary.

There are no limitations for the Business Edition except time. After about 14 days, the Business Edition will expire and you will be asked to purchase a license (you may choose to downgrade to (install) the free version instead).

Unable to auto-detect public IP and Using Private IP suggests errors suggest there may be a firewall blocking the phone. That is further supported by the fact there is a 400 Bad Request error which indicates to me the soft-phone is having trouble communicating with the VoIP server. I'm not sure what to suggest though since I don't have a compatible mobile device to test the software on.

Any TAPI compliant voice modem should work fine. Keep in mind though voice modems offer an acceptable quality, whereas a professional voice board offers a substantially higher quality. If this is a medium-large business set-up, it is recommended that you consider purchasing a voice board. Note there is a significant difference in price, a voice modem costs about $30 USD whereas a voice board can cost upwards of $200. Having said that, Voice Modems are a very convenient, inexpensive, and are great for quick install/test set-ups for your planned IVR.

What error are you getting? For the set-up: ProxyUserName -> SIP Number / Username ProxyUserPassword -> Password SIPCallerID -> Friendly Name (optional) ProxyDomain -> Server

Yes you could use IVM. If you wanted it to ring your phones (after passing the blacklist check) you could set the OGM to transfer to your Axon ring group.

Axon does not support this feature natively.. however I think there may be a way to accomplish what you are trying to do. If you had a softphone set-up as an extension that could check the caller ID against a blacklist and then quickly answer and hang-up if the number is on the list... then any incoming call on that ring group would also ring the soft phone and if the number had been black listed the call would immediately be answered and disconnected. If you wanted, you could add a delay of 1 second to the other extensions in the group so that a black-listed caller wouldn't cause the other phones to ring at all (where as without a delay, a black-listed caller would still ring once). The problem is, I do not know of any soft phone with that type of black-listing capability.

IVM has never supported this feature in the past, but from what I've heard the latest version of IVM may include support for it. It's probably best to contact NCH support directly about this and see if they can provide more info. Good to know, I'll try and remember that since timed drop-outs seem to be a somewhat common issue around here[.

Perhaps unlike some PBX systems, Axon and IVM work together to form a complete IVR-PBX. i.e. Axon simply routes calls based on pre-defined rules, if you want to prompt the user and perform an action based on input you would have to use IVM. The basic way of doing this is to set-up an IVM extension that is answered by an OGM which prompts the user to enter an extension. The OGM should be set-up in data entry mode and set to accept 3 digits (assuming all your extensions are 3 digits long) and then set to store the input into a variable "num". Then the OGM should initiate a transfer to that extension (using the stored %num% variable as the destination).

There seem to be a lot of users having difficulties with audio on PDA/Windows Mobile devices. Unfortunately I have yet to encounter a solution and I don't have a compatible device to do testing on. I'll try and keep you posted if and when a solution becomes available. You can also try contacting NCH directly to see if they have any information regarding audio problems on PDA devices.

It may be possible if you run them in separate user accounts that you keep logged on simultaneously. The only problem I can foresee would be if uplink uses global/pc registry keys instead of account based registry keys to store settings. If that's the case each time you configured one Uplink, the other(s) would also configure themselves with the same settings. Another possibility is to run another Uplink in a virtual machine. A virtual machine basically emulates an entirely different computer inside another computer. The VM (even though it is running on another operating system) acts entirely independently of the host computer. This would allow you to run Windows inside a VM and run Uplink on that. This would not interfere with the host computer's operating system or Uplink. It is essentially a way to split your computer up into two. The only problem is a VM takes a lot of resources to run (lots of processor and memory required) and I very much doubt you would be able to run more than 1 VM successfully, even just 1 VM may slow everything down to the point where audio starts to break-up.

By this do you mean that after rebooting the problem is temporarily solved and then develops again?

Shared Line Appearance (also known as BLF/Busy Lamp) is a feature that is only supported by very few PBX systems. As far as I know, Axon does not yet support this feature.

This behaviour is a bit strange. The SPA942 should have 2 line appearances enabled by default which would let you switch between 'virtual' lines 1 and 2 at any time. Are you sure Axon is actually receiving the second call? i.e. Does your VoIP line support more than 1 concurrent/simultaneous inbound call?

Are you sure you are performing the transfer correctly? In Express talk, you should first press the transfer button, then enter the number to dial/transfer to followed by pressing the little arrow beside the dial textbox (i.e. where the call button is). You shouldn't press the transfer button again after already initiating the transfer.