pythonpoole

You can only answer from extensions which are either part of the group that has been called, or if they've been called directly. Notes: A single extension can belong to several groups - Not 100% sure what you're asking, but perhaps that helps? Or perhaps you want to transfer the call to another extension outside the group?

www.nch.com.au/kb/10063.html - Here is a list of recommended Voice Modems for use with IVM (also contains supported features info) [Voice Modems: cheaper solution, lesser quality, lesser features] www.nch.com.au/hardware/telephony.html - Here is a list of recommended Telephony Boards [Telephony Boards: more expensive solution, higher quality, more features]

You can set the device for ringing in Express Talk's options window. You can also choose the sound for the ring in the settings found under the "other" tab.

Try playing around with the settings in Outbound Auto Dial Calls dialogue window. You can change how thew answer is detected (eg hardware / software) and determine how the detection of tones or voices on the line determine the start and end of the call. If you still encounter problems, come back here. It could also be that your VoIP provider acts like a proxy of sorts, where you make a call through your SIP account and instead of connecting you to the end caller directly, the service connects you to the voip centre and then from there the call is connected to the third party. So in a sense, it could be that the call is being answered straight away, and then after it's ansewered it is being forwarded on by your service provider to the caller on the other end. I haven't heard of any service that does that, but it is possible. A similar problem exists when using a Sipura FXO adapter, the call is counted as answered as soon as you begin dialing on the FXO line.

If you're talking about my guide, the answer is select any extension or group. They removed the "none" option in 1.20. Basically just so long as you understand that the setting is independent to the extension/group you set in the FXO adapter configuration.. then you're good to go. So basically to sum it up, chosing which extension or group to ring on in the line properties has no effect using my method and it will ring on the extension or group set in the FXO adapter.

Have you followed the instructions/guides for setting up the SPA-3102 with Axon? I have written one of the guides (check the pinned topics in this forum), and there is also one by NCH. If you have successfully setup the FXO adapter as described in these guides, you should have the ability to 1) Accept incoming calls from a physical exchange connected to the 3102 Adapter 2) Make outgoing calls to a physical line (same as above) connected to the 3102 adapter In terms of the dialing plans, just make sure the default line is set to the FXO line you setup while configuring your 3102 adapter earlier. After this is done all calls made from a softphone registered with Axon with that dialing plan should automatically attempt to go through the physical line attached to the 3102 adapter, and all incoming calls should be forwarded on to the appropriate softphone extensions as set in the line's properties. If you have any further questions or problems, don't hesitate to ask. Please include your SPA-3102 and Axon configuration if possible. (SPA configuration can by opening the config page in a browser and saving the webpage (it will automatically save all tabs), Axon configuration can be saved using the backup-feature in the main window).

I think I've heard the BroadVoice lines only work with Axon now. If that's the case it should be easy to have your IVM setup through Axon. It's just a case of installing Axon, configuring the line in Axon's control panel and then have incoming calls sent to an extension that is registered with IVM.

Most of the voip service providers listed on the recommended providers page (http://www.nch.com.au/talk/sip.html) are compatible with all NCH software. Other SIP services may or may not be compatible with IVM. I currently use FreeDigits and Faktortel as my voip providers and I haven't had a problem (good quality, Caller ID works correctly, can receive DTMF tones, etc.) Note: for Faktortel I had to change the codec settings in their online control panel to get it functioning correctly.

What VoIP service are you using? It could be that there are some incompatabilities with the service and the software.

h16h, I do not know the solution to your problem, but I do know that you happened to post in a topic that was pretty much exactly a year old (probably by accident). Hence you will probably not be able to get help from the people who posted in this topic. Just thought I should give you the heads up in case you're expecting a quick reply from them.

Did you close and open IVM again? IVM only checks for new TTS voices on startup.

A PBX can come in two forms: a PBX box (hardware), and a Virtual PBX (software). Both do basically the same thing, manage phone calls across a phone system. One of the only key differences is with a hardware PBX box, typically it could be designed to allow normal analogue telephones to have an extension with the PBX where as a Virtual PBX has nothing to connect these phones to so you are limited to using VoIP phones and/or softphones. In any case, a PBX is used to manage all phone calls, phone lines, phone queues, extensions, dialing plans, etc. of an internal phone network system. It can allow for example to have 1 VoIP line, and allow the incoming call to be sent to several extensions (individual VoIP phones). It can work the other way as well, allowing several independent VoIP phones to be registered to a central PBX to make outgoing phone calls. Traditionally your VoIP provider will only allow one SIP phone client to register with your VoIP account at a time, meaning if you have two VoIP phones, only one could use the phone service. With a PBX you can have the PBX register as the only software client, and then it will manage the calls on the internal phone network and allow multiple VoIP phones to make use of the same line. Of course there are a million and one more uses of a PBX, some include: - Ability to call individual phones in the network with an extension number - Ability for several voip phones to make use of the same voip line - Each VoIP phone can have a different 'line', so if PHONE A is in use, calls can still come in on PHONE B, and/or someone could make an outgoing call on PHONE B - Ability to setup internal automated attendant extensions (eg for collecting voicemail) - Set-up calling queues (eg technical support centre, users call in, placed in queue with music playing until someone is available.. then it rings the representatives phone extension) - Allow the use of several telephone lines from a single VoIP phone. (eg, I can make qne receive calls from my Australian Line, Canadian Line, and other VoIP lines I have set-up, all from the same phone. This way I don't have to clutter the place with different phones for each active line I have and I can take advantage of the cheapest local and international call rates) - Advanced dialing plans enable you to selectively choose which phoneline to use based on the format of the dialed number (eg pre-fix with 9 to use external line, or prefix with 1 to use north american line) - A million more uses

Is the call cut off after ~25 seconds or is the message cut off after ~25seconds? If it's the latter, you can change the message length in IVM's settings window. If the call is unexpectedly cutting off after 30 seconds, there is a more serious problem that needs to be investigated.

VoIP providers can set caller ID up to work one of two ways. 1) The provider sets the Caller ID information when sending the call to the receiving party 2) The provider obtains the Caller ID information from the client software and passes it on Preferably you want 1, thus no matter what happens on your end with Caller ID, when you call someone it will always appear as the Caller ID set with the provider Theoretically if you change the display name in Express Talk's configuration, it may send that on to the provider if they use the #2 method, however I'm not sure about that. If you change the display name and nothing happens, I believe to my knowledge that you are out of luck, and unless the provider can set it up so it sends the Caller ID information instead of trying to retreive it from Express Talk than I don't think you'll be able to solve the problem. Most likely the client they provide is set to send the appropritate caller ID linked to your account automatically, where as third party software is not.

This simply means that when taking into consideration the dialing plan, the number "103" doesn't go anywhere. Even though you may have a "103" extension, lets say in your dialling plan you have it set to call on your voip line as the default line for all numbers, then it will probably try and phone out 103 on to your voip line, and your voip service will simply send an error back saying that isn't a valid number / not found. Typically these are two common setups used in business/soho voip networks to workaround this: 1) Set up the dialing plan so that dialed numbers that are prefixed with a special code (eg 0 or 9) are dialed on the external voip line, and all other numbers are assumed to be internal. Pros: - Lines are clearly separate, pick which line to dial on approach, system not confused about which line to dial on. Cons: - Guests and new users to the system will have to be instructed on how to 'dial out'. - Could confuse people in emergencies preventing them from contacting an emergency response centre (eg dialing 000 instead of 0 000 or 911 instead of 9 911) - Annoyance, Inconvenience (users may forget the to add the prefix number at first and it may take getting used to) 2) The way I use Find a digit which no external calls should start with (for example, where I live, there are no local numbers or area codes starting with the digit 2). Assign all calls prefixed with that number to be dialed on the internal line, all other calls go on the external line by default. Pros: - Normal dialing as with any common PSTN line, no need to 'dial out' - No confusion in crucial situations such as emergencies or with guests - You can tell users their extension numbers with the digit prefix included. This way when they are dialing, they will not need to remember the prefix, but rather the whole number like '2198' even though the actual extension is 198 Cons: - Works on the assumption that external calls will not start with the chosen digit, otherwise users may try to call a number (eg 2123 4567) but find it won't go through because it is trying to look for it as an internal extension

I believe the default username and password is admin (for both of them), but you can always change it in the program's settings.

I am having the same problem. I already contacted NCH about this and it seems they only experienced the problem in Vista (I'm using XP and having the problem). They also said they are waiting for the manufacturer (Yealink) to release a new Vista Compatible driver for the USB phone, perhaps it will solve the problem. If not, and it is indeed true that the phones changed in 2007, this may explain the problem (as mine are new as well).

I think he may be referring you Axon (free software from NCH) which is basically a virtual call management system. It allows for semi-complex outbound dialing rules (eg if a number starts with this digit, use this line and add a 1 in front of it, basically customizeable to however you need it to be. Axon integrates well with IVM so they can work as 1 unit for answering and transfering calls and such, it's just that IVM alone has limited control over such things as call transfers, actions to take in case of errors (such as if the line is busy) which line to use, time till answering a call etc.

If it can't detect your public IP try enabling the stun servers and/or the upnp option in settings. If both are these are still on and you're experiecing problems, perhaps your router's firewall is blocking network connections to the program and you should try portforwarding the appropriate ports on your router if you haven't already done so. If none of the above works, well there is probably something wrong, but you can always resort to specifying the IP address manually if necessary.

When using my method the error will continue to appear every once in a while. Using the 'official' method, you can avoid the error, but I have yet to be able to set mine up that way and have it work as intended. Currently my setup is working fine allowing me to call out and receive calls through the line connected to the Sipura adapter. I don't really mind the message and just leave it and it doesn't seem to have a problem.

In the Sipura config page under Line 1 settings (in Admin > Advanced mode) set the 'PSTN Ring Thru Line 1:' option to No. This will still allow calls from Axon to ring the phone connected to the FXS port, but when a PSTN call comes into the FXO port it is not directly routed to the phone connected to the FXS port.

Error 401 means Unauthorized. In other words, this is most likely because the username or password you entered was incorrect, or your account with your service provider has not been activated yet. Also: If you haven't already done so, you'll need to forward the appropriate ports on your router. This will allow VoIP data to easily pass through your router's firewall directly to the Axon computer, minimizing delays, loss in quality, etc. The ports you need to forward are listed in Axon's settings. See http://www.portforward.com for more details. Note, replace spam with H T T P, the forum is currently experiencing a problem where links are being prefixed with spam://

You should set it up with the ports you have IVM set to use. Usually, by default port 5060 is used for voip communication, but for example, I have set mine up to use 5063, so it's best to double check. Also the codec may be the cause of the calls dropping in and out, different codecs use different ammounts of bandwidth. Some of the default ones use about 64kbps per call (thats a lot!). Some codecs are more compressed and can send pretty much the same quality of voice but only using small amounts of bandwidth like 8 kbps. Please note IVM supports: G726 (I believe), GSM, A-law, and U-law... and does not support G723 or G729

I've sent you a detailed e-mail, hopefully we can find a way to solve your problem together. P.S. In case readers are wondering, the topic poster privately e-mailed me prior to me viewing this topic, normally I would have answered the topic here so the community could learn from the problem and the most effective solutions.

If you use IVM's default voicemail system, all messages will be recorded to a mailbox in the IVM client. These messages can be: 1) Accessed by an internal (voip) phone or if you wish, an external phone as well 2) Accessed via your intranet or the Internet (downloaded as a wav which obviously gives you complete playback control). 3) From IVM directly (built-in player gives full playback control) You can also create your own modified system that works like a tape answering machine by recording messages to a wav file of your choice (using the actions in an OGM) and choosing the option to append the next message on to the last one. This way you can play all the messages from a single file. However it is up to you to manage this file, and share it across the network with whomever needs it. You'll also probably want to create a different file for each day or you'll end up with 1 massive file with way too many messages that'd have to be sort through.