pythonpoole

"Use the default server as the outbound proxy" in Axon is the same as not specifying an outbound server in other software as far as I know, so that would be the option to select. If you are registered, and then later it won't let you register (failed).. it could be a couple of things: Possibility: Fonosip may require you to re-register every 60 seconds or so otherwise registration attempts fail (some VoIP providers do this).

Well it depends on what kind of hardware you're using. If you're just using a voice modem on an analogue line, if you pick up any of the phones on the line you will be able to communicate while the person is leaving a message. Same thing goes if you're using Sipura 3000/3102 FXO Adapter, any analogue phones on the line can be answered For VoIP lines though, I don't think this is possible. Unless you could find away to forward the call to an extension or join the call as if it was a conference call or something, then I can't see how it'd work.

There are very few VoIP services that don't use SIP to communicate, Skype is one and maybe a couple of others around the world, but in general almost all provider use SIP even if they lock the device and don't tell you what your SIP account details are. I don't know about that particular device. I do suggest that you look into the SPA 3000 or SPA 3102 Sipura FXO adapter. It will take any analogue line, whether its coming from the phone jack, or the ATA of any voip service and it will interface itself with NCH software such as Axon and indirectly with IVM so you can have incoming calls managed by the computer using Axon and have various rules about where to send the calls, eg ring the phones in the house or send the call to IVM to take a voicemail message. The advantage of the Sipura FXO adapter is it also enables you to call out on the phone line from any digital phone connected to Axon over the network. If you want more information on the advantages of the FXO adapter, what its purpose is and how it compares to voice modem / board please PM/Message me

There are very few VoIP services that don't use SIP to communicate, Skype is one and maybe a couple of others around the world, but in general almost all provider use SIP even if they lock the device and don't tell you what your SIP account details are. I don't know about that particular device. I do suggest that you look into the SPA 3000 or SPA 3102 Sipura FXO adapter. It will take any analogue line, whether its coming from the phone jack, or the ATA of any voip service and it will interface itself with NCH software such as Axon and indirectly with IVM so you can have incoming calls managed by the computer using Axon and have various rules about where to send the calls, eg ring the phones in the house or send the call to IVM to take a voicemail message. The advantage of the Sipura FXO adapter is it also enables you to call out on the phone line from any digital phone connected to Axon over the network. If you want more information on the advantages of the FXO adapter, what its purpose is and how it compares to voice modem / board please PM/Message me

There are very few VoIP services that don't use SIP to communicate, Skype is one and maybe a couple of others around the world, but in general almost all provider use SIP even if they lock the device and don't tell you what your SIP account details are. I don't know about that particular device. I do suggest that you look into the SPA 3000 or SPA 3102 Sipura FXO adapter. It will take any analogue line, whether its coming from the phone jack, or the ATA of any voip service and it will interface itself with NCH software such as Axon and indirectly with IVM so you can have incoming calls managed by the computer using Axon and have various rules about where to send the calls, eg ring the phones in the house or send the call to IVM to take a voicemail message. The advantage of the Sipura FXO adapter is it also enables you to call out on the phone line from any digital phone connected to Axon over the network. If you want more information on the advantages of the FXO adapter, what its purpose is and how it compares to voice modem / board please PM/Message me

Axon, NCH's free Virtual PBX software has many great features, especially when combined with other NCH software such as IVR Answering Attendant and IMS (Music on hold software), VRS (For recording phone calls), and Quorum (for managing confrence calls). These are just some features / possible uses of the software that I've just thought up on the spot: Basically Axon on its own can: - Accept incoming calls from any number of phone lines (may be limited to 64 not sure) - Phone lines can be VoIP or normal PSTN analogue phone lines (when used in conjunction with an FXO adapter) - Manage incoming calls and ring the appropriate phones/extensions when the call is received - You can have it ring multiple phone extensions on an incoming call so if someone is on the phone the call will still ring through the available not in use phones for available call center agents to pick up and answer - Can setup multiple rules for what happens when a user phones in. A start-prompt message can be played, you can send the caller to an interactive menu or simply have them put on hold for the next available agent. - Setup advanced dialing plans allowing users to phone out on any of the Axon connected phones on any of the set up phone lines with the possibility to alter the number based on what it starts with. Rules can be setup to dial on different lines depending on how the number is dialed - Mechanism also put in place that enables Axon to automatically use another phone line if the attempt at dialing on the desired phone line failed (eg line busy, not functioning). - Internal extensions can be called to communicate with individuals in the callcenter - Set up queues/groups/line-ups of phone extensions, so when a caller is in they simply wait in queue until someone is able to answer the call. In conjunction with IVM: - Have complete control over the phone call, what messages are played to the user, in what order etc. There are many events that may occur to which you can apply actions to - Set up interactive voice menu systems allowing the caller to choose from any number of options, do anything from storing variables, database information, executing programs, or putting them in queue for the next available agent - Set up multiple voicemail boxes for employees with the possibility of retrieving messages with access code via telephone, intranet, internet, email etc. - Setup a P/A announcement system (using voicemail & call screening feature) In conjunction with IMS: - Set up a series of on-hold messages and/or music to be played continuously in a loop. When a call is placed on hold, users can be automatically put on to the IMS music/announcement on hold system until the line is taken off hold - Play important announcements while callers are on hold so they can be informed about issues which they may have interest in while they wait. Play music to keep the callers entertainied while they wait and not feel like phone call is dead or never going to be picked up In conjunction with VRS: - Record phone calls to the hard drive for future inspection - Useful to monitor call quality and/or employee's service level when on a call - Also useful for legal evidence, or as a record to how a call actually played out if it turns into heated discussion, threatning, blackmailing etc. In conjunction with Quorum: - Manage a confrence callcenter - Enable mutiple phone lines / extensions to merge together in one conversation using this software - Could be used for meetings and/or to allow employees to phone in and listen to meetings and such: when sick, travelling or for whatever reason the employee was unable to attend As you can see there are many possibilities, these are just a small number of possibilities I thought of off the top of my head. Setting it up how you want is pretty much only limited by your own creativity. NCH's software can do almost everything a major expensive call management system does, and even if there is something the software is can't do... Software such as IVM has an SDK for you to build (or obtain) plugins for in order to expand the functionality of the program even further.

Interesting, I'll have to experiment with that method, I tried for ages to get the adapter to work with Axon with pretty much no success whatsoever. Finally through information I found on this forum and elsewhere on the internet, I managed to piece together a method that did work to an extent. Actually, everyone I contacted and found online who got their FXO adapter working with Axon did it pretty much the same way as me and also had the register error for some reason and they all complained that following the NCH guide did not get it working. The limitations of my method that I have noted so far are: 1) If caller id is turned on, Axon simply does not want to answer the phone call, but its fine if the caller id is not forwarded 2) The register error appears often When I have time, I'll look over the steps you described above and try to adjust my settings to those. If I can get it to work, I'll try and re-write / simplify some of the guide with the new NCH supported method.

Sorry, my fault, I thought this was the Axon forum so I said use Axon, but you can use Express Talk or Axon with the VoIP service, either or will do.

Great topic, this is very useful, not sure if was you, but someone was talking about the pop-up message problem just the other day.

As far as I know, the stun servers are just there to try and obtain your computer's (external) IP address, ususally you would just ignore them as they should not affect the operation with different service providers. When you set up your Peopletelecom account with Axon you will be given a number, password and a server. Your settings should be like this: Friendly Name: Anything you wish, can also be used for caller ID sometimes Username: Your username of more likely the number Peopletelecom gave you Password: The password provided by Peopletelecom Server: Peopletelecom's SIP server. Usually its in the form of sip.peopletelecom.com, but this is unique for each service so check with them / the instructions they give for the server address. If you are unable to locate some of these settings, you can try to open the Grandstream configuration page and obtain the settings from there.

Usually you don't have to specify an outbound server unless the service provider tells you it is necessary. Also it is a bit strage that the server is just "fonosip.com" usually its in the form of <subdomain>.<domain>.<ext> example: sip.myvoipprovider.com. Are you able to make or receive any calls from fonosip? Does it register? What error is given in Axon when you try to make a call.

Please realize this forum has very few members, and most of those members only come to check on the forum once in a while. So it will naturally take a long time usually for you to get answers. You should arrange a time with other people to phone. I highly doubt you would want phone calls at 3 in the morning. Also this way, if people are going to call you can make sure the computer is not turned off and that Express Talk is running. Please also realize Express Talk is primarily used in business environments to manage large phone networks that work over their computetr network. Therefore there are even less members here who have the intention of using the program to contact other people just for chats etc. The program is mostly used to connect to a central PBX/Switchboard like Axon which is highly sophisticated software for routing calls to their intended destination and to manage voicemail etc. for many (maybe hundreds or more) users. If you are looking for people to talk to over VoIP. There are many voip services that are intended just for that and offer a public directory of hundreds of people to call, and this can be done through express talk.

It is possible that the router is blocking all SIP traffic through it. Since the only thing that has changed in the router, it is more likely your router configuration that is causing the problem and not Express Talk. I suggest you contact the manufacturer of the router / their support group. If the other softphones do work then it must be your Express Talk configuration in which case it could be something like the port numbers used.

Some places to find Aussie dialtones: http://vk6hgr.echidna.id.au/sound/dialtone.wav http://www.flashden.net/item/dial-tone/230

By default, all calls will go through line one unless otherwise specified. Unfortunately I don't know of a way to specify what line to dial an address book contact out on. There is a way to work around this, although its not recommended, it requires other (free) software from NCH, and a bit of configuration, but if done it should allow you to automatically determine which line to dial on based on the number / what the number starts with etc.

I'm really sorry, I completly misread your question above and thought you were asking about changing the dial plan, not the dialtone. To change the dialtone, In express talk, click "settings" and move to the "other" tab Under ring tones, select "dialtone" and then click the "change" button Select your own personal audio file Press ok, and now your dialtone should change. @Terry, usually you would setup your Express talk phone with a VoIP service (there are also free ones), most VoIP services provide free in service calls, so basically anyone with the same service as you is free to call. Here is a good one to try out and get started with: http://www.faktortel.com.au You can signup with a free account, and use express talk to register with the account and make calls through the Faktortel service. With the free account you can call anyone with another Faktortel free account. You are also able to call 1-800 numbers in Australia free of charge with the free account, even if you are not in Australia. You can also upgrade to a paid plan, and for only $7.50 a month or something you can get your own Australian phone number, 10c untimed outgoing calls to any number in Australia, very cheap rates to around the world and a lot more.

Edit: sorry, I thought you were asking about the dial plan not dial tone As far as I know express talk doesn't have a "dial plan", its just a case of you inputting the number and Express Talk dialing it. There shouldn't be any problem with just dialing the number as normal as your VoIP provider says it should be dialed. If you want to setup advanced dialing plans, like do this if the number starts with 04, and do this if it starts with 03 and so on, then install Axon on the computer you're using express talk on and then register express talk with an extension on Axon with the chosen dialing plan.

I don't think checking the DNS/Stun settings would cause continuous use of the hard drive. Personally I have not noticed anything like you describe, however I'm thinking it could be IVM's logs. Every time anything happens in IVM (that is recorded on the little log area at the bottom of the IVM screen), it is written to a log file. So if you constantly have something going on like phone calls, OGMs, errors like problems registering with a voip service.. all these things tend to add up and could mean IVM is writing log files to the hard disk 1 or more times every minute.

Well it must be possible, (http://www.nch.com.au/ivm/sdk.html)and http://www.nch.com.au/ivm/plugins.html Nevermind, I just found the answer we were looking for! Unfortunately it looks like that means you need to develop your own plugin that will return the OGM to play next randomly. Or maybe simply setting a variable to an OGM name triggers IVM to "Go to and play..." (I haven't tested that, but I don't think it'd work, but worth a try anyway).

I've never done something quite like this before. In terms of recording this into a database, I would first take the user to an OGM where it collects/asks the user for his/her id code when he/she calls in, save that to a variable to use later. Then ask if they are logging in or out and simply save "in" or "out" to a variable like %inout% based on the user input. then you can keep it simple and use the built-in feature in the OGM properties window to write a line to a log txt file with what ever text and variables you wish (at some point you would have had to randomly select whether voice id was requested (see below) and store the boolean or yes/no in a variable). Example: Log format: %user% logged %inout% on %date% at %time% Voice ID: %voiceRequested% 1075 logged in on 2007-02-22 at 09:30:26 Voice ID: No 6724 logged out on 2007-02-22 at 17:10:14 Voice ID: Yes Alternatively you could use a plugin to interface with a more sophisticated database. In terms of the random requests to record his or her name, you would have to use the the plugin on NCH's website with random functions. However, I'm not sure how you would control which OGM the user is sent to based on the result of the random number, I haven't done that before, and I'm not sure, but it seems like it should be possible.

Yes the free support from NCH is not very good, and it will take a very long time to get a satisfactory response. Unless you opt for the paid support option, the forum is probably your best source of advice. Also, I just wanted to make it clear (because after re-reading it may have been a bit confusing), that the method I suggested above requires no additional hardware, and no extra costs. The only thing you would need to obtain is the Axon software which is completely free. Also Axon was built to work with IVM, so you can have your PSTN line and your voip line connected to the same IVM system and use the same mailboxes etc. I.e. you can have IVM answer from a modem line and a VoIP line simultaneously no problem, up to 64 phone lines can be used with IVM at any given moment.

Instead of setting up your PAP2T device to register with your BroadVoice VoIP service, you should get Axon on your computer to register with the BroadVoice service. Then you can register the PAP2T as a VoIP extension on Axon. So if you want you can have it so when a call comes in, the phone attached to the PAP2T rings (as it normally would), but the only difference is the call is now routed through the computer with Axon on it, and then you can tell axon what to do for incoming calls, example send it to IVM to answer and take a message if you don't answer or whatever you want (indefinite number of possibilities). This way you don't need a modem or any analog hardware to have IVM answer the call, and you can still set it up so the phone connected to your PAP2T rings like it normally would when a call comes in. If you need a more in depth explanation on settings etc. please don't hesitate to ask.

I believe the paragraph you have quoted is simply saying that if you do not have a VoIP service and you just want to dial someone's computer with the program, and get the receiving party to answer the SIP phone call directed to it, then you set it up as above and dial for example joe@69.56.23.57. If BBPglobal is your provider and it supports IP dialling, you shouldn't have a problem using the feature. Typically however you would use BBPglobal to phone other BBPGlobal numbers (not IPs) and landline numbers (if your account has any credit). If Express talk is registered with the service, you shouldn't have any trouble making outgoing and receiving incoming calls from that line to other BBPGlobal users and landlines (with purchased credit/phone plan)

This is very strange I thought for certain when you picked the smtp option and filled out the details that IVM didn't try to use the email client on the computer at all to send emails. You may have to contact NCH about that one. Ok, so when you attach the PSTN phone line to the modem it recognizes an incoming call, but not when you plug the standard analog line that comes out of your ATA? Are you certain that the ATA is actually receiving the calls? i.e. when you plug a phone into the ATA does it ring for incoming phone calls? If It does, this is a strange problem. It could be few things I suppose, perhaps the voltage outputted by the ATA to indicate a phone ring is not very much and does not trigger the modem to answer the call, may be the regional settings are affecting it, I'm not sure. Is your Linksys PAP2T device locked? If it is, that is too bad, and yes you'll have to use the analog method you are attempting. If not there is much easier way to accomplish what you're doing digitally without the hassle and quality loss of the voice modem.

Firstly, this forum is not very popular so yes it will take some time for people to read and respond to your question. I think thats pretty obvious as only a couple of topics have been posted since you first posted the one you are referring to. Please be patient, or purchase technical support from NCH if you need a faster answer. Secondly, NCH is not obligated to provide any support on this forum, this forum is intended for users to talk to other users about the software. So, you'll have to wait for community members like myself to respond, and if we don't know the answer your topic may not be answered. Thirdly, NCH never designed this software for use on Linux under wine as far as I know. Wine has issues, its not perfect, so when things don't work with it, its usually a case of Wine has poor support for 'emulating' that feature, and not a case of the product being defective. Also because this software was not meant for linux, you will probably not find any other users here who have run this on linux and it will be very hard to get an answer from somebody with experience with that. Finally, all I can say is I don't personally know the answer, it could be WINE's lack of support for it, it could be incorrect port numbers, it could be the receiving party cannot receive the audio that is transmitted, or it could be the software can't recognize the audio device to send the audio from.