pythonpoole

Unless the DTML tones are the frequencies of digits 0-9 # and *, it will be much more difficult to accomplish this. You can setup IVM to listen for input (e.g. set to accept 4 digits) and then based on the DTMF tones it receives from the Analogue PBX perform some sort of action.. If the tones are not the standard DTMF telephone keypad tones (0-9 # and *) then you will need to have IVM record a message/wav when the PBX calls in, then create a plugin application that can analyze the audio file and decode the tones, and then return the variable value to IVM.

Not really, unless you don't mind users changing the line settings each time. If only a few people will need this feature, you can set-up each person's sip account on a different line in each Express Talk installation. Then the person can select their line (1-6) to make phone calls. This is obviously not an ideal option, but It is the only thing I can suggest at the moment. If you really want to have this feature, you should be able to create a program that lets users log in, and based on the user, change Express Talk's settings in the registry to reflect the new SIP account settings.

Express Talk has built-in support for most open/free codecs such as G.711 (alaw/ulaw) and GSM

Yeah, most modems today are TAPI compliant. However that doesn't mean it is voice compatible for answering machines and such. TAPI is just a universal method of communicating with modem hardware to get the modem to perform actions such as dial a number, hangup, retrieve the caller ID etc. Yes, IVM does need the modem to be TAPI compliant to communicate with it, but that is only one of the requirements. If you picked up a random TAPI modem and it worked for you, you were very lucky. In order for a modem to be compatible with applications like IVM, it should say it is a Voice Modem. Data/Fax modems are not the same thing. Often Voice Modems will have a headphone/mic jack on the modem card and may also come with (primitive) voice mail answering software. They also cost a bit more than a regular modem (e.g. rough prices: basic modem ~$5-10, data/fax ~$15-20, voice/data/fax ~$30-40). I have to say, the particular modem you purchased is very unclear (based on information on their website etc.) about whether it has voice capabilities.

Although express talk supports multiple lines, I have never tested trying to make concurrent calls through the command line interface (unfortunately I am not at home right now so I cannot test what happens). I don't think it would permit more than one call, but as I said I will have to test this later. However, for what you are trying to accomplish, may I suggest using NCH's IVM software instead? It is better suited as an auto-dialer and has many built-in features to make the process easier and more integrated. Also, if you want to use different SIP lines like you are suggesting... your best bet is to use NCH's Axon software (it is free of charge). This is essentially a VoIP PBX call management system that supports features such as dialing plans.

"C:\Program Files\NCH Swift Sound\Express Talk\talk.exe" -dial "18005551234"

Last time I checked the Uplink software did not support receiving key presses. Probably because Skype handles things differently. As it is a proprietary software it probably has non-standard and non-DTMF ways of sending key presses to the receiving Skype application.

I'm not expert on the subject, but I think it has something to do with the way in which the DTMF signals are sent to your phone service provider. I believe with VoIP there are multiple ways to send pressed digits to the other party (e.g. play the actual tone over the VoIP line or send a digital packet of information that signals a key press). In that case, unless your service provider supports the method of DTMF signaling that the software uses, then it won't receive the pressed digits.

In my opinion, Axon is a high quality product. The price ($0) is also an added bonus. However, Axon is simply the base call management system, for everything else (e.g. Music on Hold, Call Recording, Auto Attendant, Voice Mail, Call Conferencing etc.) you are required to purchase additional modules/programs (such as VRS, IMS, IVM, Quorum). Groups/Queues work well, there are several options allowing you to play custom messages, music, set delay times for different extensions (e.g. ring phone 1 for 5 seconds, if not answered, keep ringing Phone 1 and also ring Phone 2) etc. I am using version 1.2 (and also helped test it) as my main PBX system (8~10 extensions). It is quite stable and has some new and very useful features. I will note however that occasionally if there is a problem with my internet connection or if a network cable is unexpectedly unplugged, Axon sometimes pops up an abnormal exception error. I usually just restart the application after that and everything seems fine. I don't know whether the program is able to accept calls during that time though. The only other thing I might be concerned about for stability is if you intend on using VoIP phone lines, which in case you are not aware become completely useless in power failures and internet connection downtimes. In that respect, they are slightly more risky than traditional land-lines. I personally use the Linksys Sipura 3000 (now 3102) FXO adapter to connect my analogue phone line and it works well. I had a few issues getting caller ID passed on to Axon, but have now solved this problem. In terms of phones, the best ATA adapter I've used is also my Sipura 3000 (It works as both an FXO and FXS adapter). However I also have an Zxyel VoIP router that hasn't caused much trouble. For phones, I recommend the Yealink [http://yealink.com] brand SIP and USB phones (NCH/Altoedge also sells their USB phone [http://www.nch.com.au/hardware/usbphones.html]), however there is no problem with using regular analogue phones with an ATA adapter. Recommended headsets: http://www.nch.com.au/hardware/headsets.html For the server computer, I recommend: - Windows XP (I wouldn't choose to use Vista, but that is up to you) - Processor: 2.5 GHZ Pentium 4 or equivalent - Memory: At least 512mb (I recommend 1 GB) - 100mbit or 1gbit Ethernet connection to switch/router In terms of realtime monitoring of calls in Queue, no such program exists (so far), however I imagine it would be possible to create a program that did so by reading Axon's logs.

Under telephony settings there is an option "Allow outbound and message calls on this line" If unticked, the line will only be used for incoming calls. If ticked, the line can be used to make outgoing calls, but can also receive an incoming call. And for your question, only one instance of IVM can be running.

Well I don't know if this will help, but if the PHP server is on the same computer as Axon (as it appears) then there is no need to use the internet plug-in or to even have a php webserver. You can execute the PHP script by setting IVM to run a C:\Program Files\PHP\php.exe as an IVM Plugin Process. For the first argument, put in the path to the php script (e.g. "C:\Program Files\PHP\myscript.php"). The rest of the arguments are for the data you want to send to the script (eg 565) In the PHP script, instead of using something like: $cart_id = $_GET["cart_id"]; Use: $cart_id = $argv[1]; Hope this helps and saves you some trouble. I haven't had any unwanted extra characters return using this method so I'm assuming it's something related to the ineturl plugin you are using.

I don't think there is any feature built-in to do that. The closest thing would be "Bring Talk to the front when ringing" You can however achieve what your asking with third-party programs. I can't think of the name of one off the top of my head, but there are certainly programs out there that let you force certain windows to the top of the order. You can also write your own program to call the Win API function to bring the window to the Talk window to the top of other windows.

Sorry Gerard, I meant to post here earlier to let people know I've already had discussions with Jorge and have started development on the IVR.

Well, NCH is unlikely to provide you with that information, so unfortunately I don't see that happening. The best you could do is maybe do about 25 passwords the "create extension, check encrypted form" way and then use those passwords over and over again for the extensions you create. Obviously it isn't the ideal situation since the passwords will eventually repeat, but it does add some form of security. The only other method I can think of is using an Auto it mouse script which basically records a series of mouse actions (movements and clicks) to do repetitive actions for you. So depending on the software you use to do this, it may be possible for example to record the action of creating a new extension in Axon, then give the program a list of user names and passwords to use and then just get it setup to create extensions one by one for you.

Unfortunately the solution ended up being to move to another provider. We weren't able to solve the poor quality problem with the inphonex lines and we suspect it is probably an issue on the provider's server end. For example limited bandwidth capabilities or a slow internet connection.

Basically what I mean is: STEP 1: Create a new extension in Axon with the password of your choosing (this will be the default password for extensions created using the automated process). STEP 2: Then check and record what the password is saved as in "encrypted form" when stored in the registry STEP 3: Now using your automated program, create several extensions by adding the appropriate registry keys and for the 'encrypted password' key, enter the same value as the one you retrieved in STEP 2. Now these extensions should have the same default password that you chose in STEP 1 and you should be able to manually change the password if necessary through the Axon web interface.

I'm sorry, IVM doesn't support pulse dialing and I can't honestly think of any solution to fix your problem. If you contact the NCH staff, they may be willing to modify parts of the IVM software for you to allow a pulse dial option (for a fee).

Sounds like a problem with your service provider rather than Express Talk. If the error returned is "Busy Here," it is equivalent to hearing the busy signal when you dial a phone on a land-line. Which would indicate either the number is actually busy, or in some cases your provider may default to sending out Busy signals when a number is dialed incorrectly or your phone plan is not active. Please check with your provider for dialing instructions. It is a common requirement for many VoIP lines that you have to call as if the destination is long distance, even if you're right next door.. meaning you often have to dial the national and/or area code prior to the number. Some information on error 486 (Busy Here):

I am not from Hong Kong, nor can I read the language on the site, but from what I read on another site... Hong Kong Broadband http://www.hkbn.net/ and http://www.hkbn.net/BBphone/offer.html may be able to provide the services you're looking for. I'm sorry I can't be more helpful. In terms of using SkypeIn, there is technically a way to connect Skype to IVM using NCH's Uplink software, but there are still a couple of 'bugs' in it that need to be sorted out. (especially to do with multiple simultaneous phone calls, and accepting DTMF input (like press 1 to do this, 2 to do that)). In terms of the questions to ask the service provider.. any service that supports the - SIP Protocol - G711 (alaw/ulaw) OR GSM codecs Should be fine.

Did you forward the necessary ports on your router / internet modem? If your router device is automatically blocking them like a firewall, Express Talk will be unable to communicate with the VoIP service. This is indicated by the line "Something is wrong with the NAT device" (your NAT device is your router). Also odd ports don't really work with RTP, and the line "It always assigns 'odd' ports which are forbidden for Rtp" indicates your router is only allowing odd ports which will interfere with your phone connection. What router are you using?

I didn't have time to look over your IP trace, but I encountered this problem once. In fact it was with the Zyxel Prestige voip modem you report being the device that works. Anyway, the problem turned out to be a codec issue, and this is probably the case regardless of the hardware. IVM only supports the free, open codecs like G711 (alaw / ulaw) and GSM. However some VoIP phones/devices are defaulted to only use certain codecs (such as G729 that are not Axon compatible) or may not be capable of supporting the compatible codecs. For me, I simply changed the setting to use G711 as the main codec and this solved the problem for me.

You can change the audio-compression settings by: - Going to the settings window - Clicking on the Audio tab - Click the large button labeled something like "Set audio compression format..." There are several formats and compression configurations to choose from.

I do not know for certain, but I do not think Axon supports this feature. It's probably best to contact NCH directly, and often they will help find a solution for you.

I think SmOke_N already established the problem is with Express talk since the hard phones (assuming they are SIP phones / connected to an ata) can talk on the same VoIP line for hours without a problem.

What version of IVM are you using? I believe IVM 4.03 might correct this problem.