pythonpoole

Well G711 uses 64kbps and that is universal standard. So if it is using less than that in bandwidth (such as the 10-13kbps), there is no way it could be using the G711 codec... that sounds more like the GSM codec. And if you're getting 5-6kbps.. there must be another codec IVM supports that I'm not aware of that uses even less bandwidth (but no doubt probably provides even less quality than GSM). Edit: Ok I checked and the other low bandwidth codec that IVM supports must be G.723.1 (when you look at it in terms of bandwidth usage). Here is a full chart with all the VoIP codecs and the bandwidth they use. http://www.voipforo.com/en/codec/codecs.php So far we know IVM supports G.711 ~ 64kbps GSM ~ 13kbps G.723.1 ~ 6 kbps

Well alaw and ulaw are both part of the G711 codec, so you really only listed 2 codecs. One of which (G729) NCH doesn't support (it is not an open or free codec). So if your VoIP provider only supports those codecs, your only option is to go with G711 which gives the highest quality but takes up a huge amount of bandwidth. Just to give you an idea, G729 is considered by many to be the "best" codec when you consider that it provides a quality close to that of G711 but for only a fraction of the bandwidth use. Unfortunately as I said this is not an option. GSM is still an ok codec, but I wouldn't recommend it.. its quality is comparable to your average cellphone/mobile phone and the audio is definitely not as clear as the other codecs. Edit: Ok I double checked what exactly the "Default settings" option does. Apparently it allows the software to determine the best codec to use and balance out quality with bandwidth.

NCH does provide a customization and software development service where (for a fee) they will add (or in your case also remove) features you want (or don't want) from their software products. For more information see this page on the NCH website.

I believe (this is off the top of my head) that the bandwidth usage is around: 64 - 68 kbps when set to high quality (G711) 13 - 15 kbps when set to low quality (GSM) So on a 256kbps line (assuming you were able to guarantee 256kbps bandwidth all the time which is very unlikely), you would be looking at an absolute maximum of 4 calls on high quality and about 18 calls on low quality. To change the quality, go into Settings > Telephony > Choose a line and click Properties > Advanced So this means to accommodate say 200 high quality calls (as you were asking about in another topic), you would need to have at least around 13 mbit in bandwidth.

I really can't see a way of doing this with Axon. I know there are some other PBX systems support a 'CID prefix' feature for different call groups (perhaps I should put this in suggested features). There are only two possible ways I can think of checking what kind of call is coming in: A: The logging method B: The theoretical probably not working method A: There is an option for OGMs to log the call. You can log the call for example with <br>SALES CALL FROM %cid% @ %time%<br> to a file called logcall.html. Now the file can be opened in a web browser. If you then set-up a web server such as Apache or Abyss, the other computers on your network can then access the page. Then on an incoming call, a person can goto/refresh the page and view what type of incoming call it is. Each OGM would have the same set-up but with different call type (e.g. support, billing, etc.). This is obviously not the preferred solution, but it is workable. B: I'm trying to think whether it's possible to write to variables like %cid%. For example if it was possible to overwrite the variable with something like SALES, perhaps when the call is transferred it will send the caller ID as SALES. I haven't tested if it's possible to write to the built-in variables but if you could and if the variable is passed on as the CID on a call transfer.. then that could be a potential solution, but in all likely hood this won't work.

Yes, you can call an SIP url/address directly if the other service allows incoming calls using this method. Most services do permit this, but some (I believe Vonage for example) don't.

Have you signed up with a VoIP service provider? Express Talk is kind of like your ordinary telephone. It will let you call other destinations if you plug it into an active and paid for phone line from your telephone company, but it's just a dead rock on its own. So in order for Express talk to actually phone other numbers, it needs to be set-up with a VoIP service provider. Here is a list of recommended VoIP services for Express Talk http://nch.com.au/talk/sip.html Most VoIP services will let you make in-network calls to other users using the same VoIP service for free and will charge low rates for actual PSTN (your regular analogue service) destinations around the world. To give you an idea of how cheap the rates are, there are some providers that offer unlimited calls to North America for just $20 a month, and other services that charge per minute at only 1c/min to destinations in North America at no monthly rate. Most providers also offer incoming phone numbers (DID) that go directly to your VoIP account. You can usually purchase phone numbers from anywhere in the world, not just limited to your local area. This is great for example if you have your parents in another state and want to let them call you at local rates using their regular telephone service. Incoming numbers vary significantly in cost, I have one that costs only $1/month for unlimited incoming, but others cost up to around $8/month.

I think the reason is for limiting the number of lines is it just isn't really possible for a computer to support that number of concurrent calls without suffering severe losses in quality or simply crashing from the overloading processor or memory use. The fact is, to support that many VoIP calls (up to 200 as you're suggesting) at once without problems would be nearly impossible without a dedicated internet connection rated at several (e.g. 24) mbit in speed/bandwidth and a computer with very fast processing speeds and lots of memory. I believe that 64 is also the physical limit for the number of lines supported by analogue devices (although don't quote me on that).

No, I don't know sorry. All I do know is the license is valid for any version that you purchased and up to any version that was released up to 3 months after that. Here is a link to version 4.02

The first thing to understand is the ATA device is completely separate to Express Talk and they work completely independent of eachother. There are many Linksys/Cisco/Sipura ATAs out there, and I only know how to configure the Sipura ones (which should be somewhat similar to the one you have). If you haven't already done so log into your Linksys ATA by navigating to its IP address in your browser. If you don't know the IP address, you might be able to pick-up the phone attached to the device and dial * * * * (keep dialing even if it starts beeping). I'm not sure if this works for all Linksys devices, but for the Linksys/Sipura 3000/3102 ATA you can use the * * * * to get into a voice menu, and then you can enter 110# to hear the devices IP address. In the web configuration page, you should see several options.. the following are the ones that need to be configured: - Proxy: textbox needs to be changed to the SIP server/domain address of your VoIP service - User ID: textbox needs to be changed to the username or SIP number you have from your VoIP provider - Password: Fill in with the password you have set with your VoIP provider If it still does not work or register properly (as displayed on the status page) it may be necessary to configure additional settings such as a STUN server, Enabling Nat Mapping and Keep Alive features, etc.

I don't believe NCH claims they 'don't have' the old versions, It's just that they only provide downloads for the newest versions of the software and recommend that you keep a backup of the installation file with your version on it. Right now I can only find the installation file for as far back is 4.02, I don't know if that is much help you (a license is valid for all the versions that are released within 3 months of purchasing). I'm pretty sure I have the 4.00 installation somewhere, but I'll have to try and find it on my old back-up drives (no guarantees). P.S. You can upgrade your license at http://nch.com.au/upgrade If you have your old license information, depending on time of purchase etc. you'll get large discounts off the the listed prices.

Well I don't have the specific instructions. The way I was using it was like so: 1) At my house, I believe I had my router set-up to forward ports 5050 to 5090 to the computer with Axon installed on it (port forwarding instructions on www.porforward.com) 2) I created an extension in Axon for the external/remote user 3) On the remote computer at the second home I also opened the router and forwarded the ports used by the Softphone on that computer. 4) I set-up the softphone to use the extension I created 5) And it worked, incoming and outgoing calls worked without problems For me it wasn't too difficult to get it to work, but I know many other people have had many problems achieving remote extensions and some users even tried to setup VPN connections to get it to work.

Yes, Axon supports incoming DID numbers. Just so long as an 'External Line' (which is the same as a 'Trunk' as some other PBX systems call it) is setup with the SIP account, you should be able to make outbound calls and receive inbound calls on your DID. However, as I said you won't be able to actually set the Caller ID (CID), it will default to what Teliax uses.

If txtlocal.co.uk provides a service where you can send txt messages to mobile numbers by simply entering sending an email to mobilenumber@txtlocal.co.uk... then you should be able to send the txt message to a number stored by IVMs variable. Make sure you are executing the email plug-in correctly. So that the 'send to' argument is equal to %email%. I don't think the send to email in the built-in voice mail settings will allow you to send to an address using variables, but I am unsure about that (not tested).

@ahoman: I don't think these forums will be a good place to find testers for your service. There aren't many people on these forums, and I'm pretty sure you're one of only very few people that has any need to call a number in Hong Kong. Most of the users here are in North America, Australia, or some parts of Europe.

First run the network setup wizard if you haven't already done so (options>network). If you still encounter the problem and you have a static IP address, you can specify the IP in the network settings tab. The other option is (if your router supports UPnP) you can enable UPnP to try and find your address). But the bottom line is, make sure you run the wizard because it'll usually automatically sort out these sort of network issues automatically. You may also have to forward the necessary ports on your router if you haven't yet done so.

I wish I could help, but I have never encountered a problem like this before. All I can say is, can you provide the status window/log in IVM reads during a call where it's being transferred, and what (if anything) shows up in the Axon status window?

Well there are two ways of doing this (both are very different though), I have used both ways so I can say for certain that it should work without problem. First Solution: If you do not already have IVM, download and install IVM (at your office). Create an OGM that tells callers to enter the number they wish to dial, then set-it-up to receive data input (the phone number) and store the value to a variable like numcall. After the number is received (e.g. after the person presses # to indicate the end of the number), set it to go to another OGM that will transfer the call to %numcall%. The result is, you phone the office up (obviously you should have a separate incoming number for this), it asks you the number you want to dial, then it transfers you to that number using the VoIP line at the cheap rates. OR Second Solution: You set-up a new extension in Axon for your home. You then forward the necessary ports on the router at the office to allow the external extension at your home access to your office's phone network. Then download and install a softphone (e.g. Express Talk) or you set-up an IP phone / VoIP ATA with the new Axon extension. The result is, you basically have an actual office phone extension, but at home.

What does IVM report in the status window when the call is dropped? Does it say whether it was a silence time-out for example, or just a regular hang-up. Are all calls dropped at ~40 seconds, or just the calls to the IVM voice mail? What if you send the caller to a 60 second OGM message for example, is the call dropped then, or only during voice mail recordings?

Yes it does have the functionality. In each OGM there is an option to select an action to take place when the OGM completes. You can select the transfer option to transfer the call to either another extension in your office (if you use IVM in conjunction with NCH's free PBX call management system, Axon), or to outbound numbers such as mobile phones or landlines (through either a VoIP line or over a PSTN line when using an FXO adapter). The same drop-down list for selecting actions is also present for digit inputs, so you can say if (for example) the caller presses 2, transfer the call to sales.

Well, one way of doing this is to have incoming calls go to an OGM where it executes a small program that adds several numbers to dial to the outbound list. The outbound list can be configured to play an Emergency message. And the result is an incoming call that triggers a program to fill IVM's outbound list with numbers to call and play an emergency message to. It is also possible to configure in the outbound list settings what happens when a call is not answered or sent to a voicemail and then execute the appropriate OGM to handle the situation.

Axon doesn't support setting the Caller-ID yet. I know this is a feature in other PBX systems, but Axon will just default to your VoIP provider's CID. Some VoIP providers let you set the CID on their end (e.g. Voicenetwork.ca through their online control panel), and other VoIP providers automatically set it to the DID number you purchased (e.g. Vonage.com). In any case I don't believe there is a work-around for this, and you may have to suggest it (although I'm pretty sure it has already been suggested) and hope it will be available in a future version of the program.

Correct me if I'm wrong, but I don't believe Uplink has a free version. There is only a registered/paid version of the program, and a 15 day trial I think. So right now I'm going to make the assumption that you have downloaded the program and are using the evaluation trial. After the trial period has ended, you will need to purchase the license to continue using the software.

Well I do know they are working on a new quick release of Axon that supports video calls, but I don't think it has been released yet, at least I haven't seen any announcement yet, and the last release hasn't been for a while. I've also heard of this problem long before a few days ago, so I'm thinking it probably is something to do with your VoIP provider. If the call is disconnecting as soon as you answered, I think it means there is a codec problem. Make sure your provider (in case it has changed in the last few days) still supports the codecs G711 (alaw/ulaw) and/or GSM. Some providers let you choose the codec to use in their web control panel.. I suggest you check if there is an option / if they've now added one.

I agree. I have heard several problems with users saying when they put their customers on hold either they get music from their Telecommunications provider or they get dead silence. I am also having problems with external calls getting dead silence when transferring with IVM (but internal extensions hear the ringing or music just fine).