pythonpoole

I have heard other users reporting similar types of problems (also with Axon).

 

At first I thought it was just a case of the browser not having session cookies enabled or something to that effect... but the issue seems pretty widespread now and I'm betting it probably is a bug in the web-server programming.

Oh I see, so you do not normally get Caller Name identification, you just want Express Talk to cross-reference the caller's number to your address book list so that Express Talk can identify who's calling by name?

 

In that case, it could just be that NCH never implemented this feature. You would need to contact NCH directly to resolve the issue.

Please clarify if by lines you mean separate VoIP accounts or number of 'line appearances'

 

Each VoIP account can carry multiple virtual 'lines' (also known as channels), however your provider may restrict the number of concurrent channels that you can use.

 

It is common for VoIP providers to limit you to 2 concurrent channels so that you can have Call Waiting. Most providers will charge for each additional channel (last time I checked it was about $30 USD per channel per month (unlimited calls on each channel) on average - this may have changed however.

 

Some providers will provide unlimited channels if you agree to pay per minute of use.

 

 

If Express Talk is limiting you so that you can only register two VoIP accounts, you should contact NCH support... however having said that, it would be very unusual for you to need so many different registered VoIP accounts. As I said earlier, each account/line can carry multiple channels (or virtual lines), the only reason why you would need to register multiple VoIP accounts is if you are signed up with many different VoIP phone service providers.

 

Please explain your situation further and I will try to help you solve your problem without needing so many VoIP accounts (perhaps the use of a PBX would help?) - as I indicated earlier, it is very unusual to need so many.

PHP can return a value to IVM using the print command.

For example, print "Hello" will send "Hello" back to IVM, although that would not do anything useful.

 

What you can return to IVM is a variable assignment.

Let me give an example:

print "MyVar=haha";

Should set a MyVar variable to haha within IVM

 

Now here is where it gets interesting...

IVM has a built-in variable called NextOGM.

If you set that variable, you can change the OGM IVM will go to next.

 

For example:

print "NextOGM=CorrectPIN";

 

Should send IVM to the CorrectPIN OGM after processing the php script.

 

If you send a variable assignment back, you can then use that variable for something else like text to speech scripts. If I remember correctly, you may need to enclose the variable name in %percent% signs when you use it in IVM.

 

I think you can also send back multiple variable assignment statements by putting an '&' in between them.

E.g. var1=101&var2=blah

 

If you need to put a space in the variable value, use '+'

E.g. var1=blue+is+a+nice+colour

 

That's pretty much it I think

This means you need to configure your broadband router to forward the ports used by Express Talk to the computer you are running Express Talk on. (i.e. your router's firewall is blocking Express Talk from making a proper connection)

 

If I can recall correctly, the default ports (which may differ for you) are

UDP 5060 and UDP 8000 through 16000

 

Refer to your router's documentation for help on how to forward ports (differs by manufacturer and model)

Does your Sipgate service provide Caller-ID with name service? If you use a different softphone, do you see the caller's name?

 

Many VoIP services will offer free Caller-ID services, but will not provide Caller Names.

This is due to the fact that the service provider usually has to pay for access to a CNAM database (for Caller Name Lookups), which is typically charged on a pay-per-call basis.

In order to save money, many providers don't offer name services or charge extra for them.

 

On a separate note, Caller-ID with name service is only available to North American subscribers, most other countries do not include the Name in the local Caller-ID standard.

IVM is a lot more versatile and can be used for all sorts of telephony applications such as custom Interactive Voice Response (IVR) systems, data entry, payment processing, etc. The ability to interface with external plug-ins allows even further application development to extend IVM beyond its intended uses. IVM is primarily designed for call-flow applications however (e.g. the push-button menus you get when you call a businesses), meaning other applications may require multiple plug-ins, some of which you may need to custom program yourself -- which can cost time & money.

 

In contrast, Dial Dictate is a specialized software that is specifically designed to assist in telephone dictation applications, meaning a lot of the work is done for you. In this way you shift some of the cost of the development to NCH which makes things easier and it also probably means you can get your system up and running faster. Depending on your project, you may end up saving more money by buying out-of-the-box ready software rather than trying to develop your application with IVM. Some of the cost of Dial Dictate is simply attributed to the fact that it is designed for professionals (legal and medical mostly) meaning NCH had to spend more time & money making the Dial Dictate platform solid and compliant with industry standards in encryption and storage of audio dictations and records. Dial Dictate also comes with the ability to interface directly with external databases to store records/dictations and this can be a really useful tool. To do this with IVM would require a third-part plug-in that may be left unsupported or limited in functionality.

 

In my personal "Einstein" opinion (according to the NCH forum rankings), IVM is more than capable of handling your project and would be my platform of choice. However, if you prefer to have most of the work done for you and have the extra built-in features available like Database connectivity, perhaps Dial Dictate is the best option. Also if you want to have a flexible and expandable system, IVM is definitely the better option.

 

My suggestion: Download both trials, give them both a try and see what works better for you, because it really does depend.

This is a problem I've had with many providers over several years.

 

It seems when someone complains about a specific service provider, they are often able to fix the problem and release a patched version, but it seems there are several other providers with the same problem and for what ever reason the patch does not fix those ones either. When I bring it up with NCH, they basically say that they haven't really experienced the problem, but if I give them a service provider as an example, they will try and fix the problem. I think NCH is missing the point here in that it seems to be a very widespread issue across many providers.

 

Don't quote me on this, but I vaguely remember someone saying a few providers send some SIP packets on a different port number (or something to that effect), and this results in Axon sending the re-register requests on the incorrect port so they are not received and eventually the server times-out and calls are no longer taken.

They should work fine together, I'm pretty sure I've used Loquendo voices at one point. Just make sure the voice you want is set as the 'default computer voice' in the Windows Control Panel. There is a bug in some IVM versions which causes IVM to ignore the voice you select and use only the default computer voice.

An OGM is an outgoing message (as you said). In IVM this includes any actions associated with each message (such as menus, plugin operations, etc.)

 

A 1 OGM license would not be suitable since that would only allow you to do one action such as greet the caller and ask them to enter a support number.

In order to ask for the phone number, verify the support number, tell the caller anything else afterward etc. etc. you would need additional OGMs.

 

Try creating the system yourself with the IVM trial version and check how many OGMs you end up using, that should help you decide on the best license.

Connecting to the stun server usually only works if the ports are properly forwarded. Also, you can still register the SIP account even if the ports are not being forwarded correctly, no audio transfer is a telling sign that blocked ports are the problem (and not having a correct IP is also a symptom for the same).

It sounds like your router or rather your firewall may be blocking the ports Express Talk is trying to utilize. Try forwarding 5060 and 8000-16000 UDP ports on your router and unblocking the same on your PC's firewall - assuming one is enabled.

 

You can also try specifying a static IP in the network settings should the problem persist.

Just to clarify, what do you mean by there is no dialtone? The dialtone is used for indicating that the line is ready on a traditional telephony system. A dialtone on a softphone is usually reserved for calls placed through a USB phone handset; it is solely for user comfort and is completely superficial. Dialtones are generally seen as an annoyance rather than a necessity for softphones when used with regular headphones/speakers and hence the dialtone is not usually standard.

 

 

On the other hand, 'ringback' tones are a bit different. In some cases the service provider might choose to generate the ringback tones themselves, in which case this would be transferred as raw audio to the client as if the call was in progress. It is however also very common for service providers to simply send what I call ring packets to the client which basically tells Express Talk to generate the ringing locally. It sounds like these ring packets are not reaching Express Talk, or at least not in the format or SIP message order that Express Talk is expecting them to arrive in.

 

The other possibility is that the ringback tone .wav file has somehow been deleted from your installation or for whatever reason (e.g. file corruption) Express Talk is unable to generate the ring sound after receiving ring packets.

 

Finally, you could be facing a similar situation to what I am experiencing with my Asterisk server. My server is set to play MOH when a caller calls an outbound number. Essentially my server plays on-hold music while the caller waits for the called party to answer the phone. In this case, Express Talk fails to play the music when other ATAs, Voice Cards and IP phones do. If for whatever reason your service provider is using some sort of music-on-hold system to generate ringing sounds, that could also be the source of your problem.

 

 

It would be best if you could provide a full SIP log trace of a typical call so we can help pinpoint exactly what is causing the problem.

I have a feeling that IVM just acts as its own form a webserver which just generates the webpage HTML code along with the images on the fly and serves it back to the browser (i.e. I'm thinking the page data is actually stored as part of the executable, not in standalone files). I think (not sure) that there is a .css file in the Application Data or NCH Software folder which you might be able to use to customize the page a bit.

I have a feeling that IVM just acts as its own form a webserver which just generates the webpage HTML code along with the images on the fly and serves it back to the browser (i.e. I'm thinking the data for the images might actually be stored as part of the executable, not as standalone files). I think (not sure) that there is a .css file in the Application Data or NCH Software folder which you might be able to use to customize the page design a bit.

Express Talk is not like Skype where the software is the service.

Express Talk is like buying a phone from your local shop, it doesn't provide you with any service on its own, you need to connect it to the telephone line from your service provider.

 

It is designed to interface with most SIP based VoIP providers to allow you to make and take phone calls through the Internet.

 

So your first step is to get yourself a VoIP provider, then you can set-up Express Talk to make calls using that VoIP provider.

I haven't used IVM in a while, but I'm pretty sure attachments were an optional setting. There is currently no way of sending a link to the message however. Instead, you could simply tell users to go to the IVM web interface to check their messages.

NCH maintains a list of compatible SIP providers here:

http://www.nch.com.au/talk/sip.html

 

Be sure to check the "more info" link on each listing however, because although they are all listed as 'compatible', you will find that many of them have some quirks or incompatibilities with NCH software like not being able to transfer calls or not being able detect key presses, etc.

Unfortunately there are no 'universal standard' settings. Tthe ports you need to forward are determined by your Axon settings, not your VoIP provider. Axon will tell the VoIP provider what ports to use to communicate with, the router has to be instructed to allow the provider to use those ports to talk to axon and vice versa.

 

How you go about forwarding the necessary ports will vary significantly from router to router, even routers of the same brand. You will need to look up how to do it in the manual or contact the router manufacturer. As I said though, the ports you need to forward are the ones that are setup in your Axon settings (for example, a common port list may include 5060 UDP and 8000-16000 UDP).

Did you forward both TCP and UDP type ports or just one type? If just one, which one?

 

Are you sure you don't have a software based firewall on your PC which could also be causing ports to be blocked?

I noticed that, during an incoming call, Express talk shows me the caller id twice in the pup up bubble on the line button.

Is that normal?

It is normal depending on where you are located. In Canada or the United States, one line should have the person's name and the other should show the number. In other countries, the caller's name is not part of the caller ID standard so you will simply see the phone number twice.

 

As for your main question, I think I have found a suitable alternative to DNIS (since Axon lacks support for this AFAIK)

 

Under advanced settings for each line in the Axon web config, there is a setting that allows you to prepend a character to the caller ID (I'm not sure if it's just restricted to numbers, but it would work best if you could use letters as it would make it easier to distinguish from the phone number). Simply enter a different character for each phone line and save the settings. Now when someone calls in, the first digit in the caller ID should indicate which phone number they have called.

It sounds like you need to set-up port forwarding using the router's web configuration utility. The router's firewall is likely blocking some of the ports Express Talk is trying to use to communicate and this can cause several different problems including failure to identify public IP address, failure to achieve two-way audio, poor quality/crackling audio, etc.

Well I hope everything works out for you.

 

Unfortunately this happens quite often for many of the telephony apps. I have personally recommended to NCH several times that they set-up a proper private and public beta testing program and have the software tested thoroughly before it becomes a final release, but I don't believe such a program has ever been implemented.

This type of problem is almost always related to firewalls blocking access to certain ports.

 

You must setup your firewall for port forwarding with the ports you have configured Axon to use. Bare in mind that your broadband router probably has an integrated firewall in addition to the one on your computer, so both must be set-up to allow IVM to transfer audio back to Teliax.

Curious as to why you are depending on IVM when you have Asteriisk functionality available????

To be fair, the native IVR functionality built-into Asterisk is not that easy to program, certainly nothing like setting up call flow in IVM. Of course you are right that Asterisk would provide a much more powerful platform for handling automated inbound/outbound calling, but for the novice, I wouldn't recommend it.

 

Anyway for whatever reason Asterisk does not find "02124551111" to be a valid extension or phone number. Make sure you double check your dialplans and see if there are any dialplan settings which differ between your IVM extension and your softphone (also consider the softphone might have its own dialplans). Also where are you located? The number "02124551111" does not seem to conform to any common world dial plans that I can think of.