Web PABX Usage
From Net2MAX
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1. Web PABX Features | |
Contents |
1 Voice Greeting
Voice Greeting is a name (with NO spaces) make up of only alphanumeric characters (a to z, A to Z, 0 to 9). This name identifies a particular list of phone numbers to transfer a caller to.
Callers are directed to different Voice Greetings (and thus will be transferred to different lists of phone numbers) depending on who they are and what time they called.
The Voice Greeting name is also used as the name of the Voice Message file to be played to greet callers before the callers are transferred to their list of phone numbers.
So, if you have a Voice Greeting named "Public" and if you want a voice file to be played then you should have a voice file called "Public" stored in the system. To record such a voice file in the system simply use the special Record function mentioned below e.g. Record:Public
1.1 Caller Input
Caller Input specifies the digits that a caller will need to press in respond to listening to the Voice Message file. If there is a match then the caller will be transferred to assoicated list of phone numbers.
A typical Voice Message will be "To talk to sales please press 1, to talk to support please press 2, to listen to the latest price updates please press 3 and to leave a voice message please press 4"
By default, if the caller does NOT enter the expected digits the first time then the Voice Message will be played two more times at 5 seconds intervals and if the expected digits are still not entered on those two occasions then the call will be terminated. (Yes, this feature is also a quick way of implementing password access control). See below for additional features now, where you can adjust how many times the Voice Message will be replayed, and the timeout between each replays.
Note: You cannot mix different caller input method within the same Voice Greeting Sequence. i.e. They are either all Skip Input, or all Input Digits etc. You can use the "goto" command to direct the call to another Voice Greeting Sequence if a different caller input is required.
Input Digits
If you choose "Input Digits" as Caller Input, you can specify a set of digits that is a combination of 0 to 9. A Voice Message file which is the same as the Voice Greeting Sequence name must exist in the File Bank for this to work correctly. When the Caller enter the digits specified, the call will be processed by the corresponding command.
NEW: If you specify "?" as the Input Digit, then if the correct digits were not received within the timeout and retry periods, the call will be processed by the associated command in that line.
Play Voice
If you choose "Play Voice" as Caller Input, then after playing the Voice Message file there is NO need to get digit input from the caller. The call will be processed directly by the associated Command. A Voice Message file which is the same as the Voice Greeting Sequence name must exist in the File Bank for this to work correctly.
Skip Input
If you choose "Skip Input" as Caller Input, there is NO playing of the Voice Message file and NO getting of digit input from the caller. The call is immediately processed by the associated Command.
Web Digits
If you choose "Web Digits" as Caller Input, it can only be used with the "www" command. A Voice Message file which is the same as the Voice Greeting Sequence name must exist in the File Bank for this to work properly. The caller can enter a set of digits when he/she calls, and these digits will be sent as $keys variable to the web page specified in the "www" command.
Note: The difference between Input Digits and Web Digits is the former requires the caller to input a set of predefined digits, the latter can accept any digits entered by the caller
1.2 Dial Order
Dial Order specifies the order of phone numbers on the list to transfer the caller to. Starting with Dial Order equals 1, when there is NO ANSWER or when a phone number is BUSY then the caller will be transferred to the next Command on the list (2, 3, 4 and so on). The dial order number must be sequential, i.e. if only order 1,3,4 are present, then the call will stop after order 1.
If a phone number with a lower dial order is answered then the Commands on higher dial order will NOT be processed at all.
It is possible to have phone numbers with the SAME dial order. That means they will all be rang at once and the caller will be transferred to the first phone to pick up. This is very useful in situations where waiting to ring one phone number after another is too long for the caller.
Note: If the phone numbers on the same dial order are not defined to use the same rate, either via the Web Dialer or the dialing prefix, the Backup (013) rate will be used for all the numbers on this dial order. Internet numbers (899 and 898) follow this rule as well.
1.3 Commands
Phone Number is the actual phone number in either full international format or local dial format, based on your Webdialer setting. The default is full international dial format. For example, 61292176000 for a Sydney landline number and 61410311070 for an Australian mobile number. To ring Internet phones you have to use the country code 899 (e.g. 89906012345) and to ring other Web PABX you have to use country code 898 (e.g. 89806012345).
Available Commands:
agent: which add, remove or check agents in a predefined call queue. Caller is normally required to enter a valid set of agent number and password to add or remove an agent. Agent number and password should be a valid Universal Number (1CC number) and the corresponding password within the Net2Max platform. Since some Universal Numbers can be quite long, you also have the option to define a shortcut in the Webdialer Quickdial section, where an Universal Number can be mapped to a shortcut. As a result, the caller can enter the shortcut number as the Agent Number, and the password of the corresponding Universal Number as the password.
e.g. agent:90112121-sales
checkmessage: which transfers caller (you) into your message bank and enable you to listen to available messages after you have entered a password e.g. checkmessage:
dial: which dials the specified number. e.g. dial:89906090112020
dtmf: which will send the DTMF digits specified to the caller, use 'w' for a 0.5 second pause. e.g. dtmf:123w
echo: which echoes the voice back to the caller. There are two types of echo you can choose on the phone. Select 1 for voice delay test, which will echo what you said straight back to you, so you get a feel of the delay. Select 2 for voice quality test, which will record your voice for 10 seconds and then play it back to you, so you get a feel of the quality of the voice. e.g. echo:
goto: which jump to the user specified greeting_id. The format is goto:xxx where xxx is the name of the greeting_id e.g. goto:demo
leavemessage: which transfers caller into your message bank and allow the caller to record a voice message for you e.g. leavemessage:
mail: which records caller Voice Message to a file and then send that file by email to an email address. The format is mail:xxx where xxx is any valid email address e.g. mail:john@hotmail.com
message: which will process the Smart Message you specified. e.g. "message:(E61290112121) This is a test message from Webpabx to Email"
monitor: which starts a recording process, any voice initiated by the commands following this will be recorded until the call is hangup. You need to specify the filename prefix (Do not include - in the prefix) for the recorded files, then select to use either an identifier number or the timestamp to be appended at the end, and it will be available for download under File Bank. The file format can be .wav or .gsm. The .wav format provides better quality and compatibility with different media players, but the file size can grow quickly and exceed your File Bank quota easily; The .gsm format on the other hand is much smaller in size with reasonable voice quality. e.g. monitor:liverecording.gsm
NB: The monitor command is only available after you subscribe to the extra "WebPABX - Monitor" function.
play: which plays a Voice Message to the caller. The format is play:xxx where xxx is the name of the voice message file. If the name is blank then the name will equal to the key that caller input e.g. play:hello
read: which will process the text you input or the return of the URL you specify via our text to speech engine, and play back the processed file. e.g. "read:this is a test."
readmail: which will read out the emails in your Group Contacts via a Text to Speech engine. e.g. readmail:
record: which records a Voice Message. The format is record:xxx where xxx is the name of the Voice Message. e.g. record:hello Note: if the name is blank then the name will equal to the key that caller input e.g. record:
ring: which plays a ring tone for upto 60 seconds to the caller, without answering the call. e.g. ring:30
room: which transfers caller into your audio conference room. e.g. room:
queue: which transfers caller into a call queue you define. e.g. queue:90112121-sales:200
saydigits: which read out a set of digits you specify. e.g. saydigits:12345
screen: which is similar to the "dial" command, except the caller will be prompted to record their name before the call is bridged to the callee. After the callee listen to the recording done by the caller, he/she has a choice to accept or reject the call. e.g. screen:61292176000
vxml: which will process a VXML page. Only a limited set of VXML tags are supported at the moment. e.g. vxml: http://wap.net2max.com/
www: which enables a remote application or database server to control the webpabx. e.g. www:http://www.yourcompany.com/test.php? In this example, WebPBAX will call the URL http://www.yourcompany.com/test.php?&keys=xxxx&ani=xxxx&dnis=xxxx&sessid=xxxx, and expect a valid WebPABX command in return. You can also specify a WebPABX command directly here, 4 variables: $keys,$ani,$dnis and $sessid will be made available to the command, i.e. www:read:$keys, will read out the digits entered by the caller.
1.3.1 "dial" and "screen" command options
Ring Timeout allows you to specify for how many seconds the number will ring before it goes to the next dial order. The default and maximum duration is 60 seconds. e.g. dial:30
Announcement File gives you the option to play a voice file to the callee after the call is answered, but before it it bridged. The voice file must exist in the File Bank, and the file extension (.wav or .gsm) etc should NOT be included here. If the file name starts with the character "_", then after the voice file is played to the callee, the call will be bridged straight away. If the file name does not start with "_", then after the voice file is played, the callee will have an option - "Press 1 to accept the call, or 9 to reject the call". It is recommended a fake ring tone or music is enabled together with this option, that way the caller will hear the ring tone or music rather than silence while the voice file is played to the callee. e.g. dial::_sales
Fake Ring Tone allows a fake ring tone or music to be played back to the caller. Sometimes it may take longer for calls to be connected to mobile or overseas numbers, during which time it will be silence on the line only. With this option turned on, the caller will know the call is progressing immediately, rather than wondering what is going on during the call connection. If you defined customized Hold Music, then this music will be played if you choose Music as the fake ring tone. e.g. dial:::m
Transfer Permission with this option enabled, caller or callee can press *1* (Blind Transfer) or *2* (Attended Transfer) during the current call, to transfer it to the third party. Although transfer is mostly available with SIP or IAX phones without this option, this option will help in those calls connected to mobile or PSTN network where transfer is not normally available. e.g. dial::::tT
Record Permission with this option enabled, caller or callee can press *3* during the current call to start recording of the conversation. The recorded file will be saved to the File Bank. e.g. dial:::::Ww
Send DTMF to with this option, you can specify DTMF tones (0-9,A-D,*,#) to be sent to caller or callee before the call is bridged. "w" can be used here as a 0.5 seconds delay. e.g. dial::::::D(123w5:57w)
Call Status Check this option is disabled by default, which means if the call is not answered for any reason, it will goto the next dial order; if the call is answered, Web PABX will disconnect the call after the current call is hung up. If this option is enabled, Web PABX will try to detect the status of the call, and further processing is possible depending on the status. There are 6 status available: ANSWER, BUSY, CANCEL, CONGESTION, CHANUNAVAIL and NOANSWER. If any of these status is detected, Web PABX will look for a "special" Voice Greeting name ${CURRENT_Greeting}-${Status}, e.g. Public-BUSY, Public-CANCEL. If the Voice Greeting name is found, it will look for the sequence with Caller Input "Skip Input" and dial order "1", when found, it will process from there on. If the "special" Voice Greeting name is not available or the Caller Input is something else, the call will continue to the next dial order in the current Voice Greeting sequence. e.g. dial:::::::g (note that 'g' is used if Call Status Check is to be enabled)
Call Limit with this option enabled, you can specify the total duration (Limit Total) for this call, and an optional time left (Limit Warning) to play a warning message, as well as how often to repeat that warning message (Warning Frequency). Once the total duration is reached, the call will automatically be disconnected. Optionally, you can upload 3 files with specific file names to your File Bank, in either .wav or .gsm format:
limit_connect: This will be played to the caller once the call is answered.
limit_warning: This will be played to the caller when the "Warning" time left is reached, and every "Warning Frequency" thereafter.
limit_timeout: This will be played to the caller when the total duration is reached.
e.g. dial::::::::L(1800000,60000,6000) This will set the maximum call duration to be 1800 seconds, and play a warning message when there are 60 seconds left, and repeat this warning message every 6 seconds until the call is finished. Note that large figure is due to the unit here is in ms, even though we translate from seconds to ms automatically for you on the web page, you should always x1000 if you use the import function.
1.3.2 "mail" command options
Copy to Conference if this option is set to Yes, the voice mail will not only be delivered to the email address, it will also be copied to your Online Conference, so you can play it while in the conference. e.g. mail:Y
System Instructions if this option is set to No, then our system prompt "Please record your message after the tone. To finish, press the hash key" will NOT be played, only a beep sound will be played. This allows the caller to have a customized greeting for the voicemail. e.g. mail::N
1.3.3 "queue" command options
Queue Name here you can define the name for the call queue, your account number will automatically be prefixed to the queue name in case different users using the same queue name. e.g. queue:90112121-sales
Queue Timeout with this option you can define the timeout period of the call queue. After this period is up, and nobody pickup the call, Web PABX will go to the next dial order. e.g. queue:90112121-sales:300
Hold Music with this option you can choose whether the caller hears the hold music or standard ringing tone while in the queue. e.g. queue:90112121-sales:300:r
Announcement here you can specify the name of a file in the File Bank, it will be played to the call agent as soon as he/she pick up the call. You cannot define this from the import function.
Member Timeout this is the ringing timeout period for each agent in the queue. If the agent does not pick up the call within this period, it will try the next agent in the queue. You cannot define this from the import function.
Member Retry after all the agents are contacted, but still nobody pick up the call. You can define here how many times you wish to try the agents again. You cannot define this from the import function.
Strategy here you can choose how the queue contact each agent. You cannot define this from the import function.
ringall - ring all available agents until one answers.
round robin - take turns ringing each available agent.
leastrecent - ring agents which was least recently called by this queue.
fewestcalls - ring the agent with fewest completed calls from this queue.
random - ring random agents.
rrmemory - round robin with memory, remember where we left off last ring pass.
1.3.4 "agent" command options
Agent number if a phone number is entered here, based on the available drop down options, this phone number can be added or removed as an agent to the queue, with no voice prompts played and no authentication required. It can also be used to check whether an agent is already logged in. If the agent is checked to be not logged in, Web PABX will look for a special Voice Greeting name ${CURRENT_Greeting}-noagent, i.e Public-noagent. If the Voice Greeting name is found, it will look for the sequence with Caller Input "Skip Input" and dial order "1", when found, it will process from there on. If the "special" Voice Greeting name is not available or the Caller Input is something else, the call will continue to the next dial order in the current Voice Greeting sequence. e.g. agent:90112121-sales::61290112121:check
System Instructions if this option is set to No, then instead of the system default sound files, you can define customized sound files for the agent command. You can upload or record them to your File Bank with the following specific names:
agent-user: This will prompt the caller for a agent number.
agent-pass: This will prompt the caller for a password of the agent.
agent-incorrect: This will be played when the agent number and password do not match.
agent-options: This will be played after the agent is authenticated. Agent will have 4 options: 1.Login; 2.Logoff; 3.Change agent phone number or 4. Exit.
agent-loginok: This will be played when the agent press 1 to login.
agent-loggedoff: This will be played when the agent press 2 to logoff.
agent-newnumber: This will be played when the agent press 3 to enter a new answering number.
agent-newnumber-confirm1: This will be played after the agent enter the new number, normally it will be something like "You have entered".
agent-newnumber-confirm2: This will be played after the agent hears the confirmed number played back, and it should be something like "Press 1 to confirm this is correct, 2 to reenter the number".
agent-newnumber-ok: This will be played after the agent accept the new number, and the new number will be updated.
e.g. agent:90112121-sales:u:61290112121:check
NB: To change new answering number for an agent, you must setup Remote Update Number 1 in the WebPABX Voice Greeting section. The new phone number will be updated using "dial" command with no options enabled.
2 Default Settings
By Default there is a sample Voice Greeting called "Public" already set up for you, which has the values Key Pressed = s and Dial Order = 1 and the Phone Number = that of your Internet Phone number.
When a call is directed to the "Public", the playing of the greeting voice message file will be "skipped" and the call will be transferred to your Internet Phone immediately.
If your Internet Phone is BUSY or there is NO ANSWER then it will go to Dial Order = 2 which then takes a voice message from the caller and send it to your email address.
You are of course welcome to change the default setting to suit your own needs, by putting in your own Voice Greetings and associated Key Pressed, Dial Orders and Phone Numbers.
2.1 Voice Greeting
Voice Greeting Mapping
| Called Phone Number | Caller Phone Number | Voice Greeting |
|---|---|---|
| All | 61290112121 | admin |
| All | 899060290112121 | admin |
| All | Available | public |
| All | NotAvailable | public |
Voice Greeting: public
| Caller Input | Dial Order | Command |
|---|---|---|
| Skip Input | 1 | dial:899060290112121 |
| Skip Input | 2 | leavemessage:u |
Voice Greeting: admin
| Caller Input | Dial Order | Command |
|---|---|---|
| Skip Input | 1 | checkmessage |
3 Change Log Details
Quality Improvements On New Network
OZtell is building a separate "simple" without baggage VoIP network from June 2006. The first phase got turned on about 3pm (Sydney, Australia) today, people receiving calls from and making calls to normal phones will experience much more consistent quality.
To test out the new quality, people can use the ECHO command from:
(1) Internet Phone - dial 0# as destination number to access the Member Menu then press 7
(2) Access Number - dial 0# as destination number to access member Menu then press 7
(3) Web PABX - use the echo: command (for advanced users)
Echo will allow you to see which part of the call is giving you quality problems.
Details on the Member Menu is available here: http://en.net2max.com/Access_Number_Usage#Member_Menu
We got 3 more phases to go with this new network. But the first phase we put in today is significant in fixing up those random quality problems that some have complained about before.
Each phase will take about 2 to 3 weeks, we expect each phase to improve both quality and reliability substantially.
New SIP and IAX Servers
Network Improvement PHASE 2
As part of phase 2 of call quality and reliability improvements ( mentioned here: http://forum.net2max.com/index.php?showtopic=2377 ), we have new SIP and IAX servers online!
They are at
sip2.syd.net2max.com
iax2.syd.net2max.com
To use them, you need to
1. Change your SIP or IAX server setting on your VoIP device from "sip.oztralia.com" or "iax.oztralia.com" to "sip2.syd.net2max.com" or "iax2.syd.net2max.com"
2. Inside www.net2max.com set it to go to "sip2.syd.net2max.com" or "iax2.syd.net2max.com" under your SIP or IAX pages.
They should have better quality for VoIP to VoIP calls than sip.oztralia.com and iax.oztralia.com (esp. when BOTH VoIP phones have moved across). We will be tuning them to have better VoIP to PSTN quality than sip.oztralia.com or iax.oztralia.com over the next 2 months.
Note:
1. "sip.oztralia.com" and "iax.oztralia.com" will still be there and operate as before. And you can move back to them at anytime if required. Their new names are "sip.syd.net2max.com" and "iax.syd.net2max.com" on the SIP and IAX pages.
2. You will also see a "sip.hkg.net2max.com" server on the drop down list, it is in Hong Kong. You can use it if you are in Asia to get better call quality.
New Web PABX Features for Callee
There are now two new features that will help the callee (the one who answers the phone) handle incoming calls much better:
1. Announce Voice File to Callee
Allows the callee (normally the Net2MAX Member) to listen to a specific voice file BEFORE speaking to the caller. This allows the system to announce a specific message to the callee (for example, the message may be some information about the nature of the incoming call).
This feature is also great for message announcements or broadcast, like dialing a lot of retail shops to tell them about a price rise of a certain product or the changing of status of a certain situation.
For details click link below: http://en.net2max.com/Web_PABX_Enhancement..._File_to_Callee
2. Local Number Override
Allows the callee (normally the Net2MAX Member) to see which phone number the caller has called, on the incoming Caller Line Identification (CLI) display on their phone. Without override, the callee will see the phone number that the caller has "called from"; but with override, the callee will see the phone number that the caller has "called to" instead.
This override feature is great for seeing which phone number the caller has called BEFORE answering the call. Note, not all call rates will pass the CLI, so pick a call rate that support the passing of CLI (e.g. Premium Rate).
For details click link below: http://en.net2max.com/Web_PABX_Enhancement...Number_Override
Remote Computer Control
Remote Computer Control of Web PABX is available from today. This enables any application or database server on the internet to control Web PABX operations in real-time based on its own live operational data.
Under Voice Greeting definitions, whenever "www" is defined as Caller Input, the input digits will be passed onto an external web site (e.g. http, https) and then the result will be used as the Command for Web PABX.
In the Command field the destination web site should be specified after the "www:" command. The Web PABX then will append 3 pieces of information to it and submit to the member's web site:
1. Input Digits (keys=)
2. Caller Phone Number (ani=)
3. Called Phone Number (dnis=)
4. Session ID (sessid=)
Examples:
1. If you have the following step in your Web PABX
www 1 www:http:// www.yourcompany.com/test.php? then Web PABX will call web page
http:// www.yourcompany.com/test.php?keys=xxxx&ani=xxxx&dnis=xxxx&sessid= to get the resulting Command to execute
2. If you have the following step in your Web PABX
www 1 www:http:// www.yourcompany.com/test.php?yourreference=xxx& then Web PABX will call web page
http://www.yourcompany.com/test.php?yourreference=xxx&keys=xxxx&ani=xxxx&dnis=xxxx&sessid=xxxx to get the resulting Command
Valid Commands:
All current Commands on Web PABX can be returned by the Member's own web site and executed:
1. checkmessage:
2. dial:
3. echo:
4. goto:
5. leavemessage:
6. mail:
7. message:
8. play:
9. read:
10. record:
11. room:
12. screen:
the only exception is the www: command itself which should not be returned by the remote computer.
Applications:
1. Allows different Commands to be used under different situations in real-time based on the Member's own arbitrarily complex logic and processing.
2. Allows Web PABX to collect data from the caller and then pass it back to the member to get appropriate responses (like reading out account balances to a specific caller through the text to speech function of the "read:" command).
3. Allows Web PABX to control remote programs and devices easily (like turn on or off the air conditioning in a house when a phone call comes in).
4 898
For advanced users (not recommended for the general public), it is possible to dial the Web PABXs of Net2MAX Members directly.
Destination numbers for automatic answering devices (Web PABXs) are just like everyday phone numbers (with country code, area code and a unique number for that specific device within the area).
The country code is always 898 for automatic answering devices on the Net2MAX platform. The area code is a 3 digit number that will change depending on which phone network the member's account is (e.g. 060).
For example:
- Destination 89806090112121 is the Web PABX of the member with account number 90112121 in area 060.
- Destination 89806090111010 is the Web PABX of the member with account number 90111010 in area 060.
Besides transferring callers to the SAME normal phone number as the entered Verified Destination Number, it is possible to transfer the call to a Web PABX instead.
Extra Ways of Checking Voice Messages
Introduction
The member menu (inside Access Numbers and Internet Phones) has been enhanced to include the checking of voice messages.
http://en.net2max.com/Access_Number_Usage#Member_Voice_Menu
There is no need to put in checkmessage: in your Web PABX anymore (if you don't want any fancy filtering and mapping).
Usage
You just need to access the Member Menu (dial 0# as destination phone number) from your Internet Phone ( SIP, IAX or Web Phone) or from the worldwide list of toll-free Access Numbers ( http://en.net2max.com/Access_Number_List ) and then select "8" as the option.
Fax PABX Released
The incredible power of the Web PABX function has now been combined with the power of the Fax to Email function to create the new Fax PABX option.
1. How it works
The Fax PABX option is available within the Access Number and Local Number functions, when selected incoming calls will answered and if a fax tone is detected (the caller is a fax machine) then the call will be transferred to the Fax to Email function automatically. Otherwise the call will be passed on to Web PABX to be processed as a voice call.
2. Benefits
You can now use ONE local number to receive BOTH voice and fax calls! Also people calling you on fax enabled Access Numbers will be able to send you a fax OR speak to you at anytime. The fax detection logic and automatic call switching is transparent to BOTH the voice caller and to the calling fax machine.
3. Notes
Since the incoming call has to be answered first by Net2MAX for fax detection to work, the caller will be be charged for the call by the carrier he or she is using - even if you do not "answer" that call with your real phone!
There is also a slight delay of 1-2 secs for the fax detection process to work.
If you do NOT want the incoming call to be answered for fax detection purpose then you have to use Web PABX instead of Fax PABX to route the calls to you.
Web PABX In-Call Menu Released
You can now press 3-digit key sequences while on the phone to tell Net2MAX to perform operations for you during your call.
- 0* Terminate Current Call
- 1* Blind Transfer Current Call
- 2* Attended Transfer Current Call
- 3* Start Recording Current Call
Some traditional phones and VoIP phones do NOT support In-Call features (like call transfers and recording etc.) and when they do support In-Call features, they use their own interfacing (like special buttons or key sequences) that are totally incompatible with each other.
Net2MAX now provides a member based In-Call Features which is IN ADDITIONAL to all the features on your traditional or VoIP telephone handset. It gives In-Call features to phones that do NOT have them and provide additional STANDARD In-Call features across all phones that have them.
Further Details Here:
http://en.net2max.com/Web_PABX_Features#In-Call_Member_Menu
Smart File Name for Web PABX
1. Introduction
Previously, if you define an announcement file to be played to you when you answer the phone, you have the choice of Pressing 1 to ACCEPT the call or Pressing 9 to REJECT the call, AFTER hearing the voice announcement you specified.
While this is great for Web PABX to give you audio warning when your mother-in-law is calling (you can reject all calls from her phone number easily by just listening to the specific announcement file played for her caller phone number), it is NOT very convenient for businesses who want to accept ALL calls.
For people who want to accept all calls, and simply want to use the announcement file to tell them who is calling or what phone number has been called, Web PABX now have an added feature to save time and effort for those people.
2. Smart File Name
From today onwards if the first character of your voice file is an underscore _ then the system will NOT ask you (the callee) to press 1 to accept or press 9 to reject, the call will be put through to you immediately after the announcement.
This saves the caller waiting time and saves you listening and pressing a key. The call flows much more smoothly for both parties.
Having a voice file name beginning with a _ instructs the system to skip the accept/reject prompting after playing the announcement file.
3. Example
dial:89906091132:30:sales The phone number "89906091132" will ring up to 30 seconds, when picked up, the voice file name "sales" will be played to the callee. The callee then enter 1 to accept the call or 9 to reject the call.
dial:89906091132:30:_sales The phone number "89906091132" will ring up to 30 seconds, when picked up, the voice file name "_sales" will be played to the callee. The callee will be connected to the caller immediately after the _sales voice file has finished.
Worldwide Call Back Enabled
Introduction
You can now save a lot of money by taking advantage of Net2MAX's list of worldwide Access Numbers with the new Call Back feature.
Selecting the new "Call Back" option for your Verified Origination Numbers is similar to selecting the existing "Entered Destination Number" option for those "Verified Origination Numbers".
However, instead of paying your phone provider its call rate to call an Access Number, you can now tell the Access Number to call you back on your number - using the call rate of whatever VoIP service provider you have set up in your Web Dialer.
Since Access Number will be using the cheap rates that you have defined in your Web Dialer to call you back, you do not have to pay origination call cost to your phone operator for that call.
Call Back is especially good for mobile phone calls and overseas phone calls, since they are normally more expensive.
Usage
1. The phone number you are calling from MUST be a Verified Origination Number. e.g. you call from your mobile +61410123456
2. Ring one of the worldwide Access Numbers from one of your Verified Origination Numbers. e.g. Dial 138813
3. You will hear a free call message telling you that call back is now active, just hang up. You will NOT be charged for the call. e.g. "Wait for call back, this has been a free call."
4. Receive a call back form Net2MAX e.g. system calls you back on +61410123456
5. Enter the real destination phone number you wanted to call e.g. 85212345678
This feature is ONLY available if you login to http://net2max.com (it will NOT be available on http://oztralia.com)
Currently only International Dial Format (Web Dialer) is supported, within 2-3 days both International and Local Dial Format will be supported.
World's First Instant Telco
1. Introduction
You can now become you own telco in seconds! ... by charging the people that use your Shared Web Dialler with your OWN call rates!
Your users can save a lot of time, money and confusion by benefiting from your centrally maintained Web Dialler to achieve:
1. Better Price and Quality
You will find that one VoIP provider is better than others for certain call destinations at certain time. You can mix and match the VoIP providers as much as you like for your users.
2. Better Reliability and Stability
You might find that one VoIP provider is down or has poor quality. You can switch to use another VoIP provider immediately.
Many benefits to both you and your users are possible, e.g. cheaper call rates as you aggregate all the minutes of your users though your own VoIP providers.
Net2MAX takes over the troublesome money collection and micro-billing from you ... how many other organisations can collect cash for e-commerce transactions through thousands of Post Offices across the country ?
2. Usage
Previously, people using your Web Dialler to make calls ALREADY got charged by Net2MAX directly for the calls they make if you pass them through Net2MAX VoIP networks. You do NOT have to bear their call charges - as they just use your defined routes and pay Net2MAX directly themselves.
Now, this direct charging of people using your web dialer is extended to External VoIP networks that you have defined as well.
The beauty of this system is that you can set you OWN rates at any time for all the external VoIP providers you have. Since VoIP providers are coming up and going down all the time, you and thus your users can take advantage of them quickly and easily.
Previous all calls through the external VoIP networks defined in your Web Dialler are free to the Callers - you have to pay but they do not pay you. Now you can selectively charge for specific destinations through your Web Dialer.
Anyone can be a telco in seconds:
1. Login to http://net2max.com 2. Go to "Payment -> Chargeable Call -> Web Dialer Chargeable Call" 3. Enter the destinations that you want to set your own call rates.
For 3. you define 5 things:
1. Prefix e.g. 614
2. First Block Duration in seconds e.g. 60
3. First Block Price in dollars e.g. 0.30
4. Next Block Duration in seconds e.g. 60
5. Next Block Price in dollars e.g. 0.30
The above means charge 30c per minute to mobile calls in Australia
Note:
1. Call are charged when they are answered (no charge for busy, no answer etc.) 2. The call rates you defined ONLY works for calls though your External VoIP Networks (calls through Net2MAX's own VoIP networks are charged at fixed rates to your user, as before). 3. In order to prevent the build up of too much call credits inside your account. Special refunds will be available to turn your excess call credits back to cash. This refund scheme will be trialled in Aug 2007 and available in Sep 2007.
3. Charges
Beside one member paying another member for using her phone accounts, the Net2MAX system also charge the person receiving the money a handling fee.
During the current trial period there is NO handling fee:
1. Set Up Fee: $0 2. Monthly Fee: $0 3. Call Fee: 0% (with minimum $0.00)
However, in the future, we are looking at charging a handling fee for all the processing and record keeping we have to do for you, especially to people who are making money with the system (not just recovering call cost from friends and family)
1. Set Up Fee: $0 (non-profit), $29.95 (profit making) 2. Monthly Fee: $0 (non-profit), $9.95 (profit making) 3. Call Fee: 1% (with minimum $0.01)
People who intent to make profit using this system will have to pay a higher fee. Although even for profit making organisations our $29.95 set up and $9.95 is NOTHING compared to the hundreds and sometimes thousands of dollars other Service Providers charge.
We are trying to clean out all bugs and fixed the pricing before September, any feedbacks appreciated.
