pythonpoole

.. You can try searching, but I am fairly confident no such software exists. I'm not sure of the technicalities, but voice enabled modems seem to be restricted to just answering (e.g. with an automated attendant) or for voice mail, I have yet to find many other applications for such hardware. A modem can be configured to transfer calls, but only initiate a transfer on the same phone line, not to Axon itself. NCH sells the 3102 adaptor you're looking for here - for about $100 USD (Can be obtained on ebay for lower) There are also telephony boards you can buy that (in combination with the free Carousel software) can perform the same function as the 3102.. however they are a bit pricier, around $200 minimum I think.

Well I'm assuming you have set-up the extensions to be only 4 digits long? You shouldn't put the whole phone number in the extension name otherwise internal extension dialling will require entering the whole number, and also because doing so serves no purpose. The external line is set-up to route calls to the appropriate extension of your choice, thus the extension name need not contain the full phone number. In the dialling plan, you can make a rule that says something like: If number starts with 9, remove 1 digit and prepend nothing, and then dial on one of the external lines. Then you can set the default line as [EXTENSION] so that all calls are assumed internal unless the employee dials nine (to access an outside line)

Unfortunately the reverse is also true. Unless you use some kind of FXO device (like an SPA3102 adaptor, or a telephony board with carousel installed) and 'convert the line to voip,' Axon won't be able to interface with it properly.

If it weren't for the verification image which prevents automated sending, this site would be perfect to interface IVM with (using a plugin). Again here is another site, but it also requires human verification. Essentially you need to find some sort of online service that allows you to send text messages to cell phones in Brazil, and then program a plug-in to interface and send text messages through that site. Another method is to purchase a GSM interface module for a computer that allows sending of text messages, then program a plug-in for IVM to interface with the module's program/driver.

Yes this is possible. Many providers offer free text messaging (in North America) to cellphones on their network by sending an email to a set addresss e.g. phonenumber@company.com. If this is the case for you, then you may be able to do this 1) For free and 2) Using the already built mail plug-in (or even easier, using IVM's built-in mailer for when a voice mail arrives). If this isn't the case, it'll be a bit tricker and you'll need someone to build a custom plug-in for you.

The router has a built-in firewall that will control access to each PC individually. Even if you forward port 5060 for one PC, on another PC it will still be blocked. Also, you should make sure that the RTP port setting (under Options -> Network tab) should be a high value, for example 8000 - 16000. The low port numbers are more often than not blocked by the router's firewall and this can cause problems receiving/delivering audio. Also try this, put the call on hold for a few seconds and then take it off hold. Is the audio problem corrected? Also make sure your stun servers are active, or if you prefer you can manually enter the IP address, SIP and RTP port details.

For this type of application, you will need to build or have someone build you a custom plugin. Typically, the way this would work is by having an OGM ask to input the first 3 characters, and then storing that value to a variable. Then running a plugin and sending that variable to the plugin executable. The plugin (programmed in a language like C++, PHP, etc.) will then have to have a searching algorithm implemented that can reference a database and find the matching person. The plugin could then return the matching name and number as variables, along with a command to go to the next OGM which would ask the caller to confirm, e.g. "Is Mike the person you want to call? Press 1 for yes, and 2 for no" and if the caller confirms, initiate a blind (or confirmed) transfer to the number that was sent by the plugin and stored in a separate variable. It sounds pretty complicated, but it's actually fairly straight forward. However it'll become a lot more complicated if the directory is large and there is a possibility of several matches.

It seems to me if you just enter random characters into the line settings (i.e. so it's not actually registering to a VoIP server), and then you dial something like sip:mynam@myvoipaddress.com or 500@192.168.0.30 it should dial directly to the other PC. I'm not sure why it requires you to enter VoIP account information when it's not actually required. Alternatively you can sign-up with a free PC to PC VoIP service that offers free in-network calling between PCs and then set-up each PC's Express Talk to use an account with that service.

Which instruction guide did you follow (the one on the NCH website or the one pinned to the top of this forum)? Also can you please upload your current configuration? (Navigate to the configuration page (preferably in Admin -> Advanced mode) and use File -> Save Page As...)

@DFritz, it's not that uncommon. You will find that some VoIP providers (especially for businesses) will provide an SIP VoIP line without authentication, but limit access to it from one specific IP. So even though there is no password authentication, the only people who can use the line are ones using it from a particular IP address (and typically businesses will own their own static IP address). There are also other cases where authentication is not used, although I will say they are rare and it is much more common to have a traditional SIP account set-up requiring a username and password to authenticate to the system.

Axon doesn't support non-authenticated trunks. Even if you did manage to erase the username and password and still submit / save the configuration in Axon, I'd think Axon would still try and authenticate to the server regardless. What happens if you put in a random user/password? Perhaps the server will simply ignore it / auto register and there won't be a problem?

Are you referring to analogue calls (e.g. through a modem or telephony board)? Or what is meant by calls outside VoIP? Is the problem that incoming VoIP calls hang-up after getting transferred to an Axon group (when hold chimes is enabled)?

IVM does not provide a Message Waiting Indicator (MWI) feature as of yet. If it is added in a future release, I suppose Axon will have to be updated with an interface to link each extension to a specific IVM mailbox.

For the first problem, is it not possible that the user is entering a non-existing extension and Axon then transfers to that 'extension' and then the call is dropped? As a fix, you can specify a range of values that can be accepted, and if the user enters an incorrect extension they are asked to input the extension again. As for the second question, I'm not exactly sure what causes dropped calls. I myself have only experienced them a couple-few times with VoIP and usually it could be explained by an interruption in the internet connection of some sort. I suppose it could also be caused by limited bandwidth (e.g. if several calls are competing on the same Internet connection, and that connection is not very high bandwidth, some calls could simply drop-out after not being able to successfully relay the SIP messages / audio to the server and back maybe? Is the second line you have with the same company using the same voip server? e.g. if the host name is sip.voipserv.com, is the second line also sip.voipserv.com? If not, it could indicate that the new line is using a new updated server which could be using a newer version of the SIP protocol that perhaps NCH software does not fully support (perhaps?), if it is the same I cannot think of an explanation right now.

If you could, a further explanation on somethings would be of much help, I'll try and answer what I can though: Note that the way VoIP works, only 1 SIP phone/handset or softphone can connect to a particular VoIP account at any one time (otherwise it will likely only ring the device that last registered to the server). The solution is to set-up a PBX like Axon (free software provided by NCH). A PBX will enable you to share a VoIP line across several accounts. Essentially, Axon registers to the VoIP service like a soft phone or handset and then allows any number of devices to register with itself under different extensions. Axon allows you to have full control over call routing and outbound dial plan's and such, so make sure you check it out. Do you mean your voice is not received on the other end? This could be due to a lack of port forwarding. Note that you must forward the VoIP ports used by Express Talk, your handset (and if applicable, Axon) on your router, so the device is aware that it should allow VoIP traffic to flow to and from the phones instead of blocking it (the default for routers with a firewall on). Also you should enable the Quality of Service feature on your router (if there is one) as this will help prioritize voip traffic to ensure a smooth phone call (without breaking up / quality issues). Note that this feature is usually only found on the higher end products and can sometimes be referred to as "Stream Engine" Also, just for your information the other program's purposes are: IVM - Answering attendant, can do everything from basic Voicemail to a complex business phone answering system (e.g. with menus, data entry, or even over the phone automated payment processing) IMS - Application that allows you to control and play music or announcements when you put callers on hold (works with other NCH software like Axon, Express Talk, etc.) VRS - Used to record phone calls (usually in business environments e.g. call centre where you want to record all calls that take place (as opposed to manual recording that can be done in Express Talk) Uplink - Lets you interface Skype with NCH's other VoIP software (specifically Axon) allowing you to receive incoming (and make outgoing) calls to/from Skype on your VoIP devices/soft phones. Quorum - VoIP conference centre application, pretty self explanatory used for business conference calling (fairly basic compared to the conference software used by businesses today)

Try unticking the Allow outbound and message forward calls on this line option in the modem's telephony device properties, and ticking it in the VoIP line's properties. If all goes well, IVM should be forced to transfer the call through the VoIP line, however there are no guarantees this will work (I seem to remember a user having a similar problem a while back with no solution).

Yeah, I just added that assuming you only wanted your cell to have access to the system.

In the answer OGM's properties, go to the Date/Time ID tab and tick User Number ID Validity Checking and enter your cell number in the textbox and select the OGM to 'skip' to. In the second OGM's properties, under the Key Response select Data Entry (variable). Enter a variable name (e.g. numvar), set the maximum digits to the max number of digits to accept (e.g. you could set this to 10 to prevent most international calls). Under # or end select Transfer call and enter %numvar% in the textbox. I recommend a blind transfer for this application, but you can select a confirmed transfer if you want. Now just so long as one of your telephony devices / voip lines has Allow outbound and message forward calls on this line ticked (under the Options tab), IVM should be able to transfer the call. (Note that not all VoIP providers support call transfers).

From the help file: Not sure if this helps.

As of now, Axon currently doesn't provide an option to change the re-registration interval. By default Axon uses the registration interval that is sent by the VoIP provider. If none is sent, or if the interval that is sent that is not the interval their system is expecting you may experience such a problem. See if the VoIP provider is able to adjust the requested interval.

You can see how many channels are in use via the small icons in the IVM status bar. Each icon represents a line or channel (if using VoIP) and changes colour based on line status. The problem is the indicators will only let you know how many lines are currently being used in IVM, and this independent to how many channels may still be in use elsewhere on your VoIP network (e.g. after transferring the call to an agent, or someone making an outbound call). Depending on the PBX system you are using, you may or not be able to acquire info about exactly how many concurrent calls are in progress. As for calls in queue, it depends on what PBX system you are using. For example the Asterisk PBX (For Linux) can show the status of all lines, extensions and queues (I believe a plug in/additional module may be necessary), whereas Axon (the PBX software developed by NCH for Windows/Linux) does not have the ability to view the number of callers in queue or similar info (as of yet). Although there is always the possibility of reviewing the logs and using that data to determine how many calls were active at a given time and so on.

In settings call transfer, try changing !, to !,#90, under blind transfer. Then, in the OGM where you want to transfer the call, select Transfer, then blind transfer and put in the number you want to transfer to followed by the # symbol. Let me know if this works.

A DID is essentially just an incoming phone number that is routed through the internet rather than traditional analogue telephony systems. So basically any VoIP provider out there (using SIP) will be able to provide you with DID incoming numbers for your VoIP system. Now just because you're using VoIP doesn't mean you can have multiple simultaneous calls. The provider of the DID number will have control over how many concurrent channels or 'lines' are allocated to each DID number you purchase. So if you have a DID line that allows 10 concurrent channels, it means you could have up to 10 calls in progress at any given moment. Typically most providers will give you two channels by default so you can have call waiting enabled (but you can also use the second channel to take more than 1 call). Depending on the provider they may charge extra for each channel or charge you per minute on incoming (instead of free unlimited inbound) with the benefit of having 'unlimited' channels. Note: Many providers won't even have the option of purchasing additional channels, make sure you check with them first. As for IVM's limit, technically it is 64 lines. I have yet to actually determine whether this limit is just for modem/telephony board hardware or if it also applies to any VoIP DIDs set-up with IVM as well, in any case I highly doubt you'll need to handle more than 64 calls at a time.

Sorry, I was referring to the PABX lines when I said PBX. By the way, Axon can interface with CAHTA and certain other telephony boards when used in conjunction with NCH's carousel software.

You should be able to add the CAHTA board as a telephony device in IVM and then have control over the incoming lines and their OGM destination. I don't know much about the telephony boards, but I know that you should be able to interface the pbx, board and IVM without much problems. What is the exact problem you are facing, is the card not even detected by IVM when you try to add it as a telephony device?