pythonpoole

My understanding is the SIP messages are transferred using the TCP protocol (because you want to make sure the message reaches the destination even if it takes a little longer) and the audio/RTP stuff uses the UDP protocol (like multiplayer games) so a lot of streaming data can be sent quickly and if some packets get lost, misplaced or out of order it's not the end of the world (i.e. it won't crash the phone system, it'll just cause a slight audio disruption such as a crackle or dropped word in speech, etc.) Typically you want to use TCP port 5060 for Axon and have public TCP port 5060 forward to private TCP port 5060. The RTP ports (which range greatly depending on the device) usually start between UDP ports 8000 and 10000 and can use up to 10000 ports (e.g. 8000 to 18000) for transferring audio. With many other devices the range is 10000 to 20000, I do not know for certain what the default upper limit is for Axon. You can also try putting your Axon computer on the DMZ (opens up all connections/firewall disabled for that PC) and if you still encounter problems, there may be another source to the problem (such as a software-based firewall on your computer).

Since the auto-hours option does not support multiple answer/no answer times per day, the best way to do this is to create a new OGM for each time-check. OGMs have an option to do a time-validity-check on the call and decide where to send the call based on a single time condition. What you want to do is to set-up a chain of OGMs in this format: OGM checks if time is between 18:00 and 08:00, if so send to voice mail, if not send to OGM 2. OGM 2 checks if time is between 13:00 and 14:00, if so send to voice mail, if not send to OGM 3. OGM 3 checks if it is Saturday or Sunday, if so send to voice mail, if not send to OGM with main menu / forward call to agent.

To test through Axon, create a least two new extensions through the Axon web interface. Then use softphones (such as Express Talk) to register with those extensions on Axon. Next, ensure that your dialplan is configured to allow calls to internal extensions. Then use your 2 or more softphones to phone the IVM extension at the same time. If it works on both softphones, it appears your SIP provider is limiting the number of concurrent channels (this is common practice for most providers, although usually providers offer support for 2 channels so you can have call waiting/a second line appearance).

Hmm, I don't see any reason why it shouldn't work then. As for half-duplex or full-duplex, I'm going to make an (educated) guess that it refers to whether or not the modem can make use of both pairs of wires in a phone cable, or just 1 pair. A typical RJ11 phone cable has 2 pairs of wires.. the second pair is pretty much never used unless you're using some kind of device that especially uses the alternate pair for something like a 2-Line phone, and very rarely some Fax machines and dial-up/ISDN modems. Some phone cables don't even have the second pair now.. it saves on costs and as I said, it's pretty useless most of the time. Anyway, if that is what it is referring to, that is not something you should be concerned about. Perhaps try downloading a newer drivers, or do a google search for a TAPI diagnostics utility that maybe you can use to check whether the hardware has a fault or there is a bug in the driver or if the problem exists only with IVM.

Sorry Math, I could have sworn I wrote a reply to you already. I guess I forgot to actually click 'Add Reply' which often happens when I go to research/double check something and forget to come back to the post. The first thing to check is if your VoIP provider supports multiple concurrent calls. Most of strict limitations, some don't provide support at all, others provide full support. You can try to prove/disprove this being the problem by phoning directly into IVM from Axon without using freephonie/your voip service. If it works, freephonie is limiting your concurrent calls, if not it appears to be IVM or Axon (however Axon is not supposed to limit simultaneous calls to my knowledge) If according to the service you should be able to have multiple concurrent calls, it could be that IVM is limiting the number of concurrent calls it is able to answer. Please check the line properties to see how many many 'simultaneous calls' are being permitted at a time. Although unlikely, perhaps it's possible IVM has some other limitation that prevents it from answering new calls as a safety feature in case the processor becomes overworked or the memory becomes full. In such a case, what are the specs of the computer running IVM/Axon? Also make sure that you have enough bandwidth available to handle a second phone call. If you're using G.711 or the default setting that could be using up to 80kbps and I think that's x2 for both ways, where as a low-bandwidth codec like GSM only requires a small (about) 15kbps. -- Second question: The special log feature is great and should work for what you're asking. It appends new data to the end of the log file. When I use the feature, I prefer to make it an HTML browser so it's easy to open up and view in a web browser. The basic idea is to log it in a format like this filename: myfile.html log string: %date% | %name% | %ansquest1% | %ansquest2% <br> By the way be careful when you set call variables to make sure they aren't global. That would be fine for a 1 call system, but for simultaneous calls it would then start mixing the answers between callers.

IVM only works with TAPI compliant modems, and under protocols it appears your modem supports just about every standard I can thnk of except one of the most popular, TAPI. The price doesn't usually matter, although TAPI compliant voice modems usually go for a minimum of $25 a piece and tend to average out at about $35-$40 for the decent quality ones. Just so long as the modem specifies on the box that it is both TAPI compliant and voice capable/"Voice Modem" you should be fine.

Poor call quality can be the caused by several factors such as the type of internet connection you have or the amount of bandwidth available or the distance between you and the VoIP server. When there is no sound, or only one way sound it can mean the audio is being blocked by closed ports on your router or firewall. You should try running the built-in 'Network Setup Wizard' and if that doesn't help, please visit your router or firewall manufacturer's website for details on how to set-up "port forwarding" to open ports Axon uses to make phone calls.

Unfortunately Axon doesn't support this feature, but IVM does (it lets you change the font size in the status window). This is a bit of an inconsistency that perhaps NCH should standardize across their VoIP software. Note that you can always go to \ProgramData\NCH Swift Sound\Axon\Logs to read the log info (same as in the status window) and open the logs in notepad or your preferred text editor.

Please check to make sure IVM is set to "On" mode (there is a little On/Off button near the top-left of the window). Also make sure you set the OGM to answer with (beside the On/Off switch), if no OGM is selected, it won't answer. If it's on and an OGM is set, does it work when you try the simulator? If the simulator works, how many rings do you have IVM set to before it answers (under settings)?

Express Dial does support call lists and the ability to import them. If you got your call center staff to import the new list (just a couple of clicks) from whatever network server/shared drive you have the list on then all the agents could have an updated list for each day. I also wrote a plug-in for Express Dial a while ago that would automatically load the new list into Express Dial whenever the plug-in was run. It could be put on Windows scheduler (built-in to Windows XP+) to automatically import a new list every X hours for example.

Simply create an extension for each agent that will be calling out. Then set-up an outbound line for you calls to go out on (either using a VoIP provider or a FXO adaptor connected to a normal analogue phone line). Lastly create a simple dial plan to send outbound calls out to the external phone/voip line of your choice. Then you're good to go, some VoIP lines will let multiple calls go out on the same line at the same time, others will restrict the number of calls. You can also set-up Axon to use a back-up line or lines if the primary ones are busy. For your call center agents, you may want to consider using Express Talk in combination with Express Dial for making outbound calls through Axon (both are NCH products). Note: Axon does not provide predictive dialling support.

Ypu can try enabling the "Disable call activity polling" in the advanced external line settings. This fixes some issues related to dropped calls.

For some reason IVM prevents you from answering before 2 rings in general. For analogue systems, I can understand this since the Caller ID doesn't usually get pass until just before the second ring.. but for SIP there is no need for this restriction and in most cases is unwanted. A while back I think I got it to answer more quickly by playing around with the date/time restrictions and/or some tweaks to some registry keys (can't remember how I did it).

Yes, it does. The basic idea for setting this up is to first create an Extension in Axon for each call center agent. Then create a new ring 'Group' and add all the extensions that your agents are using. Assuming you've already set-up your external incoming phone Line, set that line to send incoming calls to the Group you just created. Lastly, make sure your agent's soft-phones or IP phones have call waiting disabled (or whatever equivalent feature it uses) otherwise the agent's calls may be interrupted by ringing signalling that another call is coming in. If the soft-phones/IP phones you are using have no way to disable ringing/call waiting tones for other calls that come in, consider using a silent ring tone so that an agent can see when new calls are coming in but it won't disturb the call that is already in progress. After that, it should be good to go. When someone calls in, all the phones not in use will ring and when one agent answers, the other agent's phones will stop ringing until a new call comes in. Axon does not support ring strategies however (e.g. where you ring only one particular inactive agent for a phone call, or where you ring the agent that has had the least calls for the day, or you ring an agent that has been idle the longest, etc.)

Yes, voice modems can vary significantly in quality from 'poor & a bit hard to understand, but ok for personal voice mail' level to 'full business quality IVR level' the trick is to find a higher end one for a reasonable price. I had an Intel chipset based Voice Modem that cost me about $35, the quality was great (in comparison to others I tried) and I had no problems (apart from Caller ID). The best quality can be achieved using a professional telephony voice card/board. They are much more costly however ($200+)

Where are you located? Different regions of the world have different dial-tones, and some modem hard ware / voice boards are only set to detect specific frequencies (e.g. for the North American standard tones), so that when you import the device and use it in another country, it may become unusable. Some of the more advanced voice modems and voice cards/boards have regional settings that can be adjusted using the included driver software. The regional settings should allow you to adjust the hardware to scan for the different types of tones (e.g. dial-tone, busy/engaged tone, call waiting tone) for your area.

This is dependent on the modem hardware you're using. Some modems make louder noticeable clicks when performing some operations, and others are as quiet as a mouse. Sometimes the modem has settings to adjust the audio level of the modem card's internal speaker but in cases where it is the physical off-hook/on-hook operation that is causing a mechanical sound, there won't be a way to avoid this.

Because telephone lines are limited to 8 khz, all sounds that you import will be down-converted to an 8 khz format. If you have some wave/audio editing software, you may consider recording or down-converting the file to 8 khz mono pcm before loading it into IVM and this shouldn't require IVM to make any changes to the way it sounds. But yes, 8khz is pretty universal for telephone lines and it will sound much worse in comparison to your original 22 or 44 khz recording, but at least it gives you an idea of how the audio will actually sound on the phone. The best thing to do is to record the prompts in 8khz format from the beginning, you'll find it will retain more quality. It's the same reason that many text to speech engines come in both a 22 khz (for normal speech) and an 8khz format (for phone prompts).

*duplicate post*

Did you try re-installing Express Talk? The windows version also (had) the same 'bug' and by reinstalling the free version the 'trial' period would be restarted. Kind of an annoying work around, and not sure if it works for Macs.

@Gola Ying, Skype and Axon are nothing alike in the purpose they serve.. it would be impossible to compare them. Axon/AVPBX is used to manage call routing and determine how incoming and outgoing calls should be handled. It also allows you to connect several VoIP phones or soft phones to the same VoIP lines and share them as well as offer inter-extension dialling. Skype is simply a proprietary VoIP client (soft phone) that lets you make and take VoIP phone calls through the Skype network. The only thing similar about the two is that Axon allows you to connect soft phones to itself which are similar to Skype clients in a way. Axon is more like the server that runs the whole Skype network if anything else.

What most people don't know is the analogue telephone standard doesn't just include the numbers 0-9, asterisk (star) and number sign (pound/hash). There are four additional buttons that are part of the standard: A, B, C, D. There are very very few telephones in existence today which have these buttons on them, but they are technically part of the standard and therefore IVM is set-up to detect A, B, C and D key presses. To my knowledge, the only organizations actively using A, B, C, and D buttons are some military operations where the keys are used to signal to the PBX the priority of the call (e.g. officer could enter A before the number to get first priority on outgoing calls). If I remember correctly, I think the tones are also sometimes used by some home alarm systems to communicate with the central station (improves response time since there are 16 unique tones that can be sent to signal a message rather than 12). Other than that, the A, B, C, D keys are pretty much obsolete and non-existent.

This is a user support forum, we can't do much to help you. You should contact NCH directly either here, here or even here. Another option would be to install IVM on a second computer which would provide an extra 14 days for you to try. As a last resort, you can try performing a system restore to an earlier date (for ME, XP, Vista) and running/installing IVM again.

A NAT device is usually a router in most cases which is used to sort through your internet traffic and direct it to the appropriate computer. By default most routers have a built-in firewall that will block some network traffic. If the router/NAT is blocking VoIP traffic, it will make it very difficult or impossible for VoIP applications such as Express Talk and IVM to transmit and receive audio over VoIP. The software is recommending that you set-up your router to forward/allow traffic from the SIP and RTP ports IVM & Express Talk are using. There is however no universal instruction manual to forward ports as each manufacturer has its own design specifications and ways of forwarding ports. You can find router-specific instructions here. Also make sure you run the built-in "Network Set-up Wizard" in both IVM and Express Talk, in many cases this can fix many of the problems you're having.

I think this is more a problem with the modem hardware you're using than anything else. Perhaps you should contact your modem's manufacturer and see if they have new/patched drivers that may fix the problem. @Ciao121, sometimes VoIP key presses are not detected if the VoIP provider is using an incompatible signalling method (there are about 4 different ways to send key presses using VoIP).