Jump to content

pythonpoole

Members
  • Posts

    842
  • Joined

  • Last visited

Everything posted by pythonpoole

  1. I suggest you PM nchaj, she is the Technical Support and Development Program Manager at NCH and I'm sure she will be happy to help you/inform the programmers of the problem on your behalf.
  2. pythonpoole

    Thanks!

    Am I to assume this means you no longer need the delete file plug-in?
  3. Did you ever submit the bug to NCH (using this page)? If you just posted here, it's unlikely the developers are even aware of the issue. You can also contact the Axon support team directly here.
  4. Perhaps you should consider a Fax to E-mail line? You can get them amazingly cheap, in fact there used to be a company that gave them away for free. Essentially it's just a dedicated number they set-up for you and whenever a fax call comes in, they PDF it and send it to you via email. Keep in mind faxes on VoIP are not that reliable. Unless you keep a perfect quality line, there were likely be some trouble. Every time the quality is distorted, the voice breaks up, or the jitter buffer time shifts the audio, the fax machine could lose out on the connection / have a corrupted transmission. Most VoIP devices which do support FAX or FAX passthru note this unreliability in the device's documentation. If you want a more reliable FAX method I suggest either use a dedicated FAX line from your Telco or a Fax to Email service.
  5. This could be the case if your VoIP provider does not allow call transfers or their system doesn't properly support them (e.g. CallCentric)
  6. At this time IVM lacks the ability to execute plug-ins/programs during an incoming call without actually answering. The best you can do is answer, play a short tone and then hang-up the caller. Call them back and their total call cost would only be for a one or two call. For most Telcos this would hardly be anything (many telcos operate on a free local calling and 6 second or 30 second billing intervals without flag-fall charges (connection costs) for long distance). Unfortunately, it's not like this all over the world (not sure about Italy). For example in Australia, many of the telcos round to 30 seconds or greater in addition to charging excessive flag-fall or connection charges (e.g. 60c to connect to a call, and then 20c/min thereafter). If this is the case for your country then perhaps other options should be considered. I think NCH mentioned adding the ability to execute plug-ins on incoming (unanswered) calls and after a call has hung-up in the next version but I'm not sure about that, it could have just been a user suggestion.
  7. pythonpoole

    pap2t-na

    You need to set-up a dial plan for your extensions to use, otherwise Axon may attempt (and fail) at sending out all calls to an external VoIP provider. The dial plan is relatively simple to set-up. For example if you have extensions 100-109, you can set-up a dial plan rule for numbers that start with 10 and then set them to use line "[EXTENSION]" (for internal extension calls), then you can set the default / all else fails line to your VoIP provider so any other number (that doesn't start with 10) will be put on the VoIP line. Also make sure that the extensions you have set-up in Axon are set-up in the Ring Group 701. If 701 is the default group for incoming calls, only extensions connected to 701 will ring, and judging by the errors, Axon is complaining their are none to ring.
  8. I have set-up extensions outside my local network using Axon at one time, but I only got it working after essentially disabling the Firewall on the router and opening all ports for Internet communication. This is not recommended, especially on a non-testing server. It would indicate however that you could accomplish this task by simply port forwarding and/or unblocking all the WLAN ports that are required for the remote clients to accept/make calls. I know that many other users have tried to do the same thing with their set-up and failed, and eventually concluded that using a VPN gateway was the only option. This may be something you want to consider. The advantage of a VPN is remote computers can access the main network as if they were right there, physically plugged into it, thus anything that works on the physical network should work remotely as well. The disadvantage is that all Internet communication then tunnels through the VPN. This means if people are browsing the Internet from home, it sends the request over to it's ISP which then sends it to your office/network with the PBX, then out to your ISP, back to the office, back to their ISP and back to the user.
  9. This is a rather simple set-up. Set the outbound answer OGM to have no audio/a small blank audio file and set the action to hangup when the OGM is finished. If you want to keep the Caller ID on record, simply use the log feature in the advanced tab and use something like On %date% at %time% caller %cid% phoned in.
  10. pythonpoole

    pap2t-na

    Yes. Whenever configuring a PAP2 make sure you are Admin & Advanced mode (select Admin login, then switch to Advanced View) The LINE 1 tab and the LINE 2 tab are for setting up your 2 extensions The only settings that need to be changed are: - Line Enable: Yes - Display Name: Your choice of caller ID - User ID: The username for the extension in Axon - Password: The password for the extension in Axon - Proxy: The IP address of the computer running Axon - Save settings, wait a few seconds, and you should be good to go Not that I know of. Although some VoIP providers/hosted PBX services recommend not registering two devices from the same (external) IP because it may cause undesired effects. My understanding is the router should forward the port used by each device to that device. I.e. if Axon uses 5060, and your PAP2 uses 5061, port 5060 should be forwarded to the Axon computer, and 5061 to the PAP2.
  11. Well I'm not surprised to hear that. I seem to remember one of the earlier IVM versions had a bug with plug-in execution. I'm not exactly sure on how to correct this, if anyone can provide info on how to fix plug-ins for 4.01 I will certainly try and modify it / provide instructions for running it on 4.01
  12. This is very strange behaviour. It should act the same in the simulator as it does with a regular incoming call. I just tested both the simulator and an incoming VoIP call on my end, and there were no problems, it worked as expected. What kind of behaviour is the plug-in showing? Here are some examples and possible reasons. The plug-in does not appear at all, not even the quick splash/loading screen - Indicates the plug-in isn't being run at all, even on older PCs that may not be able to handle the program (which is not the case for you), it should show the splash screen. The plug-in briefly appears with a splash screen and immediately closes itself with no warning/error (1 of 2 possible reasons) - The plug-in was not given a proper Caller ID number. The plug-in will take in the cid variable and strip it of anything but digits. If nothing is left (meaning no digits), it closes itself. This could indicate IVM is not correctly passing on the caller ID. - The plug-in can't run on this computer (not the case as it runs fine with the simulator as you said) The plug-in shows an error indicating a failure to initialize a drawing surface - Your DirectX may be out of date, the computer may have too little video memory available (unlikely as it works fine with the simulator) "ERROR in action number 1 of Alarm Event" - Indicates the path to the IVM executable was invalid / ivm.exe was not found. You can put the plug-in in the same folder as IVM and use ivm.exe as the path to make it easier for you. I just tested running the plug-in with caller ID '3332628694' to make sure the plug-in could call it back correctly.. it worked. So it's not the number that's causing the problem either. Somehow I feel IVM must be responsible, something in IVM must be causing the CID to be passed on to the plug-in only in the simulated calls. Are you using the same OGM for simulator calls and you VoIP calls, is there anything you can think of that is making your simulator tests act differently than your real tests? What IVM version are you using? The thing is it works perfectly on my end for both simulated and real calls, so I find the behaviour you encountered very strange.
  13. Call-back & Delayed Outbound Call IVM Plug-in README [DESCRIPTION] This plug-in will allow you to initiate an outbound call to a phone number after a specified period of time. You can call a pre-defined number, or have IVM call-back the last caller. By creating a delay before initiating the call, you can ensure the line is free on the caller side before calling back. It could also be used to set reminder calls. [CHANGEABLE OPTIONS] - Number to be called (can be pre-defined or set to the Caller ID) - The time until the plug-in calls the number (default 10 seconds) - Whether or not to hide the IVM window when making the call - The path to the IVM executable (if different to the default) [HOW TO USE] - In your answer OGM in IVM, go to the Advanced tab and tick "Run Exe or Run IVM Plugin" - Click the Open Run Exe or Plugin Settings button, followed by the "Add New Exe" button - Browse and select the plug-in exe file - Arguments: No. | Arg. Name | Sample use | Required | Description ----------------------------------------------------------------------------------------------- 1 | Caller # | %cid% | YES | The number to call, use %cid% for callback 2 | Time delay| 100 | NO | Delay before initiating the call (100 = 10sec) 3 | Hide IVM | hide | NO | Whether to hide IVM (default: no, use hide for yes) 4 | IVM Path |"C:\ivm.exe"| NO | Path to IVM exe (default is the default install path) ----------------------------------------------------------------------------------------------- * Arguments must be in the exact order as shown above * Use ABC if you wish to skip an argument (use default) but use the next argument - Click Ok, Close and go back to the main IVM window, test an incoming call [TERMS OF USE] ----------------------------------------------- - If using for commercial/business purposes, please donate $10 USD for each computer you wish to use this plug-in on use PayPal.com to e-mail the money to pythonpoole@gmail.com - You may use this plug-in free of charge for testing or personal/home use - This program is copyright by Benjamin Poole of Scorptek (scorptek.net) © May 2008 - Do not re-distribute without permission (email pythonpoole@gmail.com) [TROUBLESHOOT] PROBLEM: "ERROR in action number 1 of Alarm Event for alarm0..." SOLUTION: The IVM path you used was invalid, try again, be sure to include the \ivm.exe PROBLEM: plug-in quits as soon as it opens / IVM doesn't show evidence of an outbound call SOLUTION: The number you entered in argument 1 (default is %cid%) was not valid / was empty PROBLEM: "Failed to initialize drawing surface" SOLUTION: Update DirectX, reduce video memory usage (lower screen resolution or bit depth if required) PROBLEM: Time shows as -1 and then program closes SOLUTION: This is normal behaviour, -1 indicates IVM has already been instructed to call, and the plug-in is closing [CONTACT] ivm@scorptek.net OR pythonpoole@gmail.com Note: There is a small license fee of $10 USD for business/commercial use (see README) DOWNLOAD Download the plug-in here!
  14. Call-back & Delayed Outbound Call IVM Plug-in README [DESCRIPTION] This plug-in will allow you to initiate an outbound call to a phone number after a specified period of time. You can call a pre-defined number, or have IVM call-back the last caller. [CHANGEABLE OPTIONS] - Number to be called (can be pre-defined or set to the Caller ID) - The time until the plug-in calls the number (default 10 seconds) - Whether or not to hide the IVM window when making the call - The path to the IVM executable (if different to the default) [HOW TO USE] - In your answer OGM in IVM, go to the Advanced tab and tick "Run Exe or Run IVM Plugin" - Click the Open Run Exe or Plugin Settings button, followed by the "Add New Exe" button - Browse and select the plug-in exe file - Arguments: No. | Arg. Name | Sample use | Required | Description ----------------------------------------------------------------------------------------------- 1 | Caller # | %cid% | YES | The number to call, use %cid% for callback 2 | Time delay| 100 | NO | Delay before initiating the call (100 = 10sec) 3 | Hide IVM | hide | NO | Whether to hide IVM (default: no, use hide for yes) 4 | IVM Path |"C:\ivm.exe"| NO | Path to IVM exe (default is the default install path) ----------------------------------------------------------------------------------------------- * Arguments must be in the exact order as shown above * Use ABC if you wish to skip an argument (use default) but use the next argument - Click Ok, Close and go back to the main IVM window, test an incoming call [TERMS OF USE] ----------------------------------------------- - If using for commercial/business purposes, please donate $10 for each computer you wish to use this plug-in on use PayPal.com to e-mail the money to pythonpoole@gmail.com - You may use this plug-in free of charge for testing or personal/home use - This program is copyright by Benjamin Poole of Scorptek (scorptek.net) © May 2008 - Do not re-distribute without permission (email pythonpoole@gmail.com) [TROUBLESHOOT] PROBLEM: "ERROR in action number 1 of Alarm Event for alarm0..." SOLUTION: The IVM path you used was invalid, try again, be sure to include the \ivm.exe PROBLEM: plug-in quits as soon as it opens / IVM doesn't show evidence of an outbound call SOLUTION: The number you entered in argument 1 (default is %cid%) was not valid / was empty PROBLEM: "Failed to initialize drawing surface" SOLUTION: Update DirectX, reduce video memory usage (lower screen resolution or bit depth if required) PROBLEM: Time shows as -1 and then program closes SOLUTION: This is normal behaviour, -1 indicates IVM has already been instructed to call, and the plug-in is closing [CONTACT] ivm@scorptek.net OR pythonpoole@gmail.com Note: There is a small license fee for business/commercial use (see README) DOWNLOAD Download the plug-in here!
  15. Yes, this is called a callback feature. The best method is to have IVM answer with an OGM which runs a plug-in and provides it with the argument %cid% The plug-in should then wait for a few seconds and run the IVM.exe executable with -outbound <variable with caller-id number here> as the arguments Meanwhile the OGM should hang-up the call just after running the plug-in IVM will then call back the person in a few seconds (using the default outbound OGM) based on the user's caller id number IVM can do this without a plug-in, but it would mean the call would go out before the person hangs-up (if you had more than 1 line/channel). If you only had 1 line and didn't use the plug-in method, it may still work, but it could also call back too quickly after the caller hangs-up for the call back to work correctly.
  16. I think the multi-line feature is referring more to the fact you can hook up a Voice board and connect several physical analog lines to IVM. It may also be referring to VoIP lines (in VoIP each 'line' can have several channels or 'virtual lines' linked to it). I'm not exactly sure how one physical line can ever allow more than 2 simultaneous call though. To be more technical, almost every standard phone wire has the ability to carry up to two phone lines. In a normal set-up, the extra 2 smaller wires inside the phone wire aren't used. In a 2-line set-up, 2 of the wires are used for line 1 - the other two for line 2. This is how a multi-line phone usually works (makes use of all 4 wires inside the phone wire and requires 1 phone wire per every 2 lines). I'm not aware of it being possible to have more than 2 calls on one physical line without some special cable/equipment. Instead, I'm pretty sure the line would have to overflow to another physical line if the first one is busy. I'm still learning about how analog phone systems work (my focus is more on digital), but I'm pretty sure this is how most set-ups would be. One phone number would be linked to line 1 which would automatically forward to line 2 if busy, which would go to line 3 if busy and so on. Your telephone company does not face this restriction. Because of the way their equipment operates, your phone company can easily forward all calls to Voice Mail for your line without problems, even if there are several simultaneous calls. If you have an answering system on your end however, you are faced with these line restrictions, and if your lines are busy, your phone company would automatically forward calls to your old Voicemail, or play a busy tone if you un-subscribed.
  17. Strange, I have the same exact voice (as well as several others), and I've never had a problem using it with IVM. In fact, it was the default voice I used for quite a while. Does it work fine when you use a different voice?
  18. I have had similar problems to what you're describing, but not specifically with IVM (or any other NCH software for that matter). I know I have had SAPI 5 voices that simply refused to play in some TTS software, but worked fine in others (it always worked with IVM for me though). I also had some SAPI 5 voices that I couldn't get to work at all with most TTS software, after re-installing them the problem seemed to be corrected. Lastly, I've heard that some TTS voices require that you have certain licensed software installed and/or running (e.g. Text Aloud) otherwise the Voices will refuse to work. My Suggestions: - Restart the computer if you haven't already done so - Re-install the Voice & related software if possible - Try using the Voice in other TTS programs like TextAloud - Contact the manufacturers/creators of the voice and see if there are any special requirements for using the voice in third party apps. By the way, what TTS voice are you using?
  19. In Windows, it's under the File menu called Run Audio Set-up Wizard... don't know if that helps. It's quite possible it doesn't exist on the Mac version. You may also want to consider posting your question in the Mac OSX specific forum to see if anybody else has similar problems.
  20. What soft phone are you using? Express Talk and most other soft phones have a gain (volume) adjustment for both audio in & out. If you are using an external USB phone that could also be the source of the problem. If you're using Express Talk, run the Audio Set-up Wizard and see what the mic levels are. If they're low, it would indicate either a problem with the microphone/connection or even more likely the Operating System's settings for mic volume (Note that Windows also has a feature called "Microphone Boost" that lets you really turn the gain up on the mic's input significantly (can cause clipping & hissing though)). If the volume levels appear to be normal, it does appear to be a problem in the transmission. Even if it is a problem in the transmission of voice data, I would think the problem could still be solved to a reasonable satisfaction by maxing the gain-out on the soft phone, turning up the mic volume in the sound mixer, and if necessary, enabling the mic boost option. I thought I should also note that the Linux version is pretty new (as in beginning of this month) and I'm not sure how many people are actually using it in real world applications as of yet. It's quite possible that this is a problem exclusive to the Linux version and has not been reported yet.
  21. I believe the correct Port Forward mapping is to forward the SIP port used by each different extension to the particular computer/device that extension is on. I.E. if Axon on PC1 is using 5060, forward 5060 to PC1.. If Extension 102 with port 5070 is on PC 2, forward 5070 to PC2. Keep in mind though that I don't think WAN to LAN port forwarding will help much for internal calls, I think it's more to help calls to/from your VoIP provider. Note: Failure to connect to the STUN servers is often indicative of firewall port blocking / incorrect port forward set-up. You can also use UPNP as alternative. UPNP is not available on all routers, and even on the routers that support it, the functionality is sometimes limited. Having said that, UPNP is a great feature in theory that helps take all the hassle out of port forwarding and network set-ups. I don't know exactly how it works but my understanding is it allows free communication through the firewall on specific ports used by the software (auto port forwarding?). Also make sure that your internal network (LAN) is not blocking anything (or at least not any ports related to Express Talk).. if your network is for home or a small business, there really shouldn't be a need for a LAN to LAN firewall as most if not all the communication between PCs in your network would be trusted. In any case, this sounds like it could be one of the major problem(s) you're having as the difficulties you are experiencing are all internal. Lastly, it is very unusual for Express Talk to report "error returned: 401 unauthorised" and then register fine later using the same username/password. I have yet to come up with a plausible explanation to this, it's almost as if line interference is causing corruption to the SIP messages so they have to be resent several times before they get through (would also explain the delay?). Perhaps blocked ports can cause similar issues. Since I haven't come across these problems before, it's hard for me to identify exactly what the source of the problem is. But it seems to me the most plausible explanation is either blocked ports on the internal LAN and/or packet corruption possibly due to interference and/or network issues. Line interference doesn't sound right though since you mentioned calling 102 from 101 doesn't cause any problems.
  22. Usually I have a solution for everything, but this one perplexes me. Usually these types of problems are indicative of a network communication issue, but you have already forwarded the necessary ports and run the network set-up wizard; that rules out most network communication problems. It's also very strange that calling from 101 to 102 works perfectly but viceversa causes a problem. Do you experience similar problems when using another softphone (e.g. X-lite or SJPhone)?
  23. Before I can answer your question, I'm going to need a better understanding of what exactly you're trying to accomplish. From what I understand: 1) You are trying to interface IVM with an external conference system 2) You want IVM to automatically answer a series of IVR prompts to join a conference 3) You want IVM to join the conference on two separate lines and try and communicate between each line So essentially what it boils down to is: you want IVM to periodically monitor a conference system and let you know if it's not working. Is this correct? If so, here is my basic idea for accomplishing this: - Using IVM's built-in outbound dial list you can make outbound calls on command or automatically using the Windows Scheduler - IVM can run an OGM when the call is answered on the other end. This OGM should pre programmed to play a sound file that contains pauses and DTMF sounds corresponding to the IVR menus presented by the conference system. (Best to use a stop-watch and manually respond to the IVR prompts first, and then recreate that in an Audio file). *Note1: If you're unsure how to create a sound file with DTMF tones, try downloading Audacity. It is free of charge and has built-in features for creating DTMF tones and pauses *Note2: You could also record a phone call with you answering the IVR prompts manually and then have IVM automatically play that when the call is answered (this is less likely to work due to quality loss from the recording) This is where things get a bit tricky. - Both lines should now be in the Conference call if all went well. The basic idea for verifying each line is in the conference is by having an OGM that plays a series of DTMF digits (e.g. 1 2 #) and then listens for a 2 digit response + # (using the data input method) from the other line. If the lines pick up the 2 digits + #, it confirms the other line is present and it should continue to play it's DTMF sound file for a few loops to make sure the other line picks it up. If it doesn't pick-up the digits, it should continue to loop (playing it's 2 digits + #, and then waiting for the response). If after several loops it still doesn't pick-up the other line, it could indicate a problem and you could then have IVM notify you (e.g. by running an external program, or playing a sound over the computer speakers). I would imagine the loop would have to repeat several times before both lines could confirm/verify the other line's presence. The reason being that you could not guarantee each line would enter the conference at about the same time and be listening at the right time for the DTMF response. Thus looping will most likely be necessary so that even if both lines were off-timed by several seconds, there would be a period of time where it could successfully receive the digits from the other line. I should also add I'm not 100% sure how well that set-up would work. I've never set-up a system like this before. My concern is that one IVM line might actually receive the digits it sends out itself in the OGM audio. This would be a semi-disaster as IVM would just confirm with itself and not actually check if the other line is present. In order to fix this, there may have to be a 2 OGM loop. (1 OGM would play the digits to the other line, and then go to a second OGM that would listen for the input).
  24. Sorry I didn't realize you were working with VoIP. You're right, the settings I gave wouldn't have changed anything. Rather than trying to detect sound levels and tones (as with analog/modem set-ups) VoIP transfers are done by sending information to & from the VoIP server using the SIP protocol. You should note that not all VoIP services support call transfers with IVM (even if they provide 2 concurrent channels). I believe Callcentric is one where a bug in their system (according to NCH) prevents transfers from NCH software.
  25. Try using answer detect using these settings: Ringing Silence Detect Time: 4100 Voice Silence Detect Time: 1000 Ringing Or Busy Timeout: 50 Answer Detect Or Fail Timeout: 50 Assume answer machine if voice longer than: 30 This should force IVM to treat Answering Machines / Voice Mail Greetings as calls that are actually picked up by a person It should also increase the time out cut-offs to prevent IVM from prematurely aborting the transfer (due to no answer)
×
×
  • Create New...