pythonpoole

Actually there are several:

-hangup

This hangs-up the current call

-answer

Answer an incoming phone call.

-hold

Puts the current call on hold.

-offhold

Takes the on-hold call off hold. If there are multiple lines on hold, calling this multiple times will cycle through all on-hold calls.

-dtmf [key]

This sends a DTMF tone down the line during a phone call. For the "[key]" parameter, you can use any numbers from 0 to 9 inclusive, as well as * and #.

-hide

Hides the window of Express Talk.

-show

Reveals the Express Talk Window

-logon

This starts Express Talk (if it is not open already) and minimizes the program to the system tray.

-exit

This closes Express Talk (complete exit).

Ah ok. Yeah most of my calls to mobiles were just short 1-2 min calls, and I guess it was the flag fall that made them cost around 55c/min on Telstra. Just for the record though I don't and won't charge flag falls/connection fees for the Scorptek service. Unfortunately the poor exchange rate at the moment between the USD and AUD is making it very difficult to lower the price much further, if it had been a few months ago I would have probably been able to give a rate of 18c AUD. Oh well, I guess that's what to expect in an economic crisis.

-

This behavior seems strange, and if indeed it acts one way for landlines and another way for mobiles, I'm almost positive this is an issue with the VoIP provider.

It's more likely the firewall is on the originating side of your ICS connection.

e.g. If you and a neighbour share a connection for example, just because there is no firewall between your neighbour and your house, doesn't mean your neighbour doesn't have a firewall on their router that is affecting you.

-

Thanks for filling out the survey, I appreciate it.

Also, you noted in the survey that high mobile phone costs are hurting your business. I know what you mean, I lived in Australia and I found the phone costs outrageous. Especially coming from North America where it is free to call mobile phones (unless they are out-of-area in which case they count as a normal long distance call).

Would you be interested in trialling out my VoIP service for your business' calls to mobiles?

Last time I checked, Telstra charged something like 50c/min for calls to mobiles.

What if I chopped that price down to about 17c/min USD? Let me know what you think.

-

As for ET, NCH Swift Sound (the Australian Company) developed the software. I'm just a beta tester who helps out (voluntarily) on these forums.

-

As for Scorptek, yes we will be providing VoIP services to the public eventually although there is still a lot of things to sort out first including a lot of major changes to how the whole system will function. As with most VoIP providers, Scorptek will manage the connection of calls through third-party carriers which will then route the calls over the internet to the appropriate exchanges (known as central offices/rate-centres in North America) where the call is then sent along the copper to the destination.

"Unable to auto-detect public IP. Using private IP" means Express Talk was blocked by a firewall and wasn't able to obtain your public IP address, so it's only choice is to identify itself by your internal LAN IP. In order to fix this the firewall must be turned off / ports must be opened OR you must specify your external IP address in Express Talk's network settings.

Also note that in most cases it is not the firewall on the PC (e.g. Norton) that blocks connections on your network.. it is almost always a router. Even if you don't realize it, often the modems or routers your ISP provides for you have built-in firewalls that restrict your connection. These are the firewalls that need to be adjusted to open ports for express talk. For more information on port forwarding your router see www.portforward.com and be sure to use the free manual guide.

^ Yeah sorry about that, I meant 5070. It's just that the standard SIP port for most VoIP software seems to be 5060, I assume ET uses 5070 by default to avoid conflicts with other SIP devices on the network such as Axon.

06:40:38 STUN server failed: stun.softjoys.com

06:40:39 STUN server failed: stun1.faktortel.com.au

06:40:39 The internet connection could be down.

06:40:39 Check local firewall is not blocking UDP port 8000

06:40:39 Also check STUN settings or use different STUN servers.

06:40:39 Unable to auto-detect public IP. Using private IP

06:40:39 STUN server failed: stun1.faktortel.com.au

06:40:39 STUN server failed: stun1.faktortel.com.au

06:40:39 The internet connection could be down.

Also, this indicates Express Talk is blocked by some kind of firewall. Are you sure that there is no software firewall on your PC that could be causing this.

If by ICS you mean Internet Connection Sharing, it is likely that the bridge between your connection and the original sharing connection could be restricted, or even more likely that the original sharing connection itself has a firewall blocking the connection.

You can try setting Express Talk to use a static IP and ports you are sure are open, this should help eliminate the firewall issue and will probably also improve your call quality.

Hmm, I suppose this could be a firewall issue. It's possible that the signal from your provider to tell ET the call has ended was blocked by your firewall and never got through, thus keeping the connection active.

Make sure you run the network setup wizard in Express Talk. If it reports any problems, there is likely an issue with your firewall.

You should make sure your Router's firewall is set-up to open and forward the following ports to the computer running Express Talk:

UDP 5060

UDP 8000-16000

Unfortunately I can see why your provider is unwilling to refund you since most likely they were also charged for the calls while the connection stayed active. However if you really push them they may be willing to refund you some of the cost, i.e. so you essentially pay the wholesale rate they are charged (not uncommon for this to happen if you file a complaint).

I should point out that this is unusual behavior for ET and (for me anyway) calls disconnect as expected. Perhaps it is a minor incompatibility between the SIP format ET is expecting and that used by your provider (e.g. sometimes the provider decides to throw in extra messages or use incorrect formatting and such that should be ignored but ET may get confused leading to unexpected behavior.

1. I'm not really sure what you mean.. if IVM answers the call and the operator's line is the same line used for IVM.. I don't see how it is possible for an external answering machine to answer since the phone line wouldn't be ringing, right?

In any case, as a general rule of thumb, playing an audio file in an OGM that corresponds to the tones played for that digit/symbol is the same as actually pressing the digit/symbol (at least to the remote machine). So if you play the # sound on the line, the machine should detect it as a # key press.

2. Try using a short ring audio file (e.g. 4-6 seconds long), make sure it's not set to repeat. If you are still having trouble, you can try executing the wav file as a plug-in and this should run it in your default media player I assume.

Also the Ring and Chimes OGMs should actually be the same OGM. I.e. the caller selects to speak with an operator, they go to a new OGM (e.g. OGM_hold). In this OGM an audio file of loopable chimes is set and an event is created so that pressing the # key will hang-up the call. In the same OGM, an external wav file of a ringing sound is set in the advanced tab.

The result is this: Caller requests operator, hears chimes, at the same time operator hears ringing sound, operator picks up phone and presses #, IVM hangs-up and the call can proceed as normal. If there is still chimes after you press #, it could mean for some reason IVM did not receive the # signal and thus it didn't hang-up.

You can't transfer the call when the operator is on the same line that IVM has already answered.

What you can do however is set it to go to an OGM that will play an external sound wav (e.g. of a phone ringing) and then have the OGM set to keep repeating until it receives a # signal to which it responds by hanging up.

How would this work you ask?

When the caller selects to speak with an operator, they are put on hold while a ringing sound is then played over the speakers from the computer running IVM. The operator then picks up the line and presses # so IVM will hang-up and the operator can talk with caller without interruption. You can also choose to play a ringing tone or loopable chimes/music to the customer while s/he waits for the operator.

It's a bug, don't worry a new fixed version should be out very soon.

A link will most likely be posted in this forum by nchaj when it's available.

Unfortunately I don't know a way of forcing an instant answer. I know at one point I discovered a key in the registry for the number of rings, but I believe there must have been a validation check because it seemed to just reset back to two on it's own and I didn't notice any difference.

I believe the limitation is in place since Caller ID is not passed on until the pause between the first and second ring (on a traditional analog system), so IVM is forced to delay the answer in order to get the Caller ID. But I agree with you that in many cases this is unacceptable and there should be an option to disable it, especially since VoIP calls send the Caller ID at virtually the same time the call is initiated so the ring delay is completely unnecessary.

I once managed to get IVM to answer instantly while playing around with the time conditions, although I can't remember how and don't know if it would still work in newer versions.

Yep, IVM will pop-up a system tray notification by default when there is an incoming call.

Make sure you set IVM to ignore the calls though. This can either be done by setting it to 'Off' mode (which might disable the notifications, I'm not sure) or setting IVM to answer after an absurdly large number of rings (e.g. 20).

In my honest opinion though, it's not worth it to purchase an IVM license for this single feature. There are other programs out there that provide a similar feature, some of them even freeware. IVM is a very feature full program that has unlimited applications and potential in the business or even home environment, and it seems very wasteful to use it for such a simple task. It would be like buying a full business customer relationship management software suite and using it as a family address book.

I've also experienced this problem and I believe it is a bug in the latest version of IVM. What you can do is go into the windows control panel and change the default voice to the French voice and that should fix the problem. Hopefully this will be fixed in the next version.

I don't think the Basic license will do what you want, it is intended for 'Answering Machine' type applications.

I believe call transfers are considered 'outbound calls' and thus the minimum license that offers the features you need would be the 'Small Interactive System' license which allows for 2 phone lines, 70 outgoing messages / interactive menus and up to 30 mailboxes.

Sorry Oris, I would have responded earlier however I had mid-terms this week and didn't have a chance to.

As you said, "telephony card, Axon, carousel"... that's pretty much all you need to set-up your basic office call routing system. As for the routing to cell phones, do you mean for someone to manually transfer calls to cell phones, to auto-forward to a cellphones after x seconds, or to ring the cellphone(s) simultaneously with any internal extensions?

Of course if you want to set-up office extensions you will have to purchase IP Phones (or ATA adapters) and connect them to your network. After that they can connect directly to Axon and be set-up for inbound & outbound call routing.

Sorry, I should have made it clear that purchasing the portforward.com software is not required, portforward.com also provides (free) instructions on their website to set up port forwarding manually with your router.

Since Express Talk is not on the list, you should select X-Lite. Note however that they do not use the same ports exactly so you'll need to use the instructions as a guide and then change it so you forward:

UDP port 5060, and

UDP ports 8000 through 16000

to your computer's IP address.

Well the first thing to do is to run the audio set-up wizard and make sure Express Talk is using the right audio device. Assuming the problem still exists, the likely cause is a firewall between your computer and the internet that is blocking the audio.

In most cases this firewall is built-into the router or broadband modem you have. What you need to do to fix this is forward the following ports on your router to the computer running Express Talk:

UDP 5060 and UDP 8000-16000

Now run the network set-up wizard in Express Talk, there shouldn't be any errors or warnings. Make a test call and it should work. If not, there is most likely another firewall blocking the connection and this is probably software based (e.g. Norton Internet Security, Windows Firewall, Zonealarm, etc.). Make sure you configure this software to allow connections to/from Express Talk (if necessary specify the ports listed above).

If all goes well your audio problems will be fixed.

The error indicates you have a firewall blocking Express Talk, this is probably the firewall built into your router.

In order to free the connection, you should forward the following ports to the PC:

UDP: 5060, 8000-16000

For information on port forwarding, see www.portforward.com

When you have finished, run the built-in network setup wizard in Express Talk and make a test call.

Are you by any chance using time conditions to direct calls? I believe if time conditions are on IVM says (or at least used to say) 'Not Answered, IVM answering mode is off' when in fact it was on. I found that it still answered the calls anyway despite the error as directed by the preset time conditions. I thought this 'bug' was fixed though in the most recent versions although don't quote me on that.

Yeah, each extension in Axon can be set to forward to a number after a specified number of seconds. I think it should work fine if you have it forward to the URI address above.. if not, try setting an SIP rule in your dialing plan which essentially tells Axon which VoIP line you want to use to make the call. I'm not exactly sure how the feature is intended to work, but I think it means if you put proxy01.sipphone.com as the host and set it to dial on the same gizmo line, perhaps it just dials the 1747XXXXXX directly or it still dials the full URI but at least it is using the correct line and not something else (e.g. the internal extensions line).

There are two ways to connect your vonage line to Express Talk / Other VoIP devices.

1) Using an FXO adaptor (purchase this, connect it between your vonage box and your network and other VoIP devices will be able to connect to the Vonage line)

2) Purchasing the 'soft phone' line from Vonage (this can only be purchased in addition to your normal Vonage line), this soft phone line can be connected directly to Express Talk and other VoIP devices. This is definitely the easiest way, but not that cheap in the long run.

Yes, if you haven't already done so, I recommend running the network setup wizard in Axon. If it reports any errors, that could be the soure of the problem.

What could be causing it is blocked ports (the wizard should indicate if this is a problem). What you should do is forward/allow UDP ports 5060-5070 and 8000-16000 on your router/firewall to the PC running axon. Now run the wizard again and see if that gets rid of the errors. Then test some phone calls and see what happens.

Good question, I'm not entirely sure. There is however 1 setting in Axon that says it corrects problems with calls being disconnected.. I think it's called 'Call Activity Polling' but don't quote me on that, look for the feature that says it fixes this type of problem. Other than that, I'm not quite sure - post again if you still have trouble.

Last time I checked, Axon has a back-up feature in it's main console window (maybe it's under settings?). You should be able to back-up and restore configurations without much effort.

In terms of the correct ports to forward, I believe it is:

5060 UDP (or whatever Axon is set to as the SIP port, may be best to forward 5060-5070 to be sure)

8000-16000 UDP (by default this is what Axon uses to communicate, the fact you don't have 8000-8010 forwarded could be the source of the problem since I believe these would be the vital ports necessary.

Also if you are still having trouble, Axon has a couple of advanced options that usually fix problems with 1 way audio. E.g. the "Always relay audio feature" and one of the other features (can't remember name) that says beside it that it corrects 1-way/no-way audio issues.

I guess it makes sense. Always relay audio would imply that the communications are forced to go through Axon before they go between the service provider (pennytel) hence any on-hold commands would be handled locally whereas before the call would be connected directly between the Axon extension and Pennytel. Maybe.. just a guess.