Ooma Telo
Page Contents
The Ooma Telo is a simple, plug-and-play solution for accessing residential VoIP (Voice over Internet Protocol) service.
The Ooma Telo is a VoIP adapter, which enables residential users with a high-speed Internet connection to access inexpensive VoIP phone services. With Ooma Telo VoIP, residential Internet customers have access to:
The Ooma Telo VoIP home phone system also offers extension service with Ooma Telo softphone apps for Android and iPhone mobile phone users.
How does the Ooma Telo phone system work?
You might think of the Ooma Telo as a base, much like a cable modem. The Ooma Telo is an analog telephone adapter (ATA) that connects to an Ethernet cable. You then connect your telephone to the Ooma Telo VoIP adapter. You do not have to connect your computer to the Ooma Telo VoIP home phone system.
The Ooma Telo VoIP home phone system functions as more than an ATA because of included telephony features. In addition to converting your analog phone signal to a digital phone signal, the Ooma Telo phone system includes free inbound and outbound caller ID, call waiting, and voicemail. Ooma Telo Premier subscribers enjoy the benefits of a second line, 3-way conference calling, voicemail-to-text transcription, and more.
Ooma Telo Calling PlanFor international calls, the Ooma Telo phone system offers two options: standard rate and plan rate. With Ooma Telo’s standard rate option, there is no monthly fee or commitment, and calls to 61 countries are usually lower than 1¢ per minute. The Ooma Telo plan rate offers 1,000 minutes of international calling for under $10 per month.
There are a few exceptions to free local and long distance calls in the United States. With the Ooma Telo VoIP home phone system, 411 calls will cost .99¢ and be charged to a prepaid phone card. Also, 900 numbers and other phone chat service numbers are not supported.
Vocal clarity with Ooma Telo’s PureVoice HD
The Ooma Telo VoIP home phone system also claims to offer better clarity than Ooma Telo VoIP competitors with its PureVoice HD technology. The Ooma Telo uses HD voice (double the voice signal), encrypted calls, advanced voice compression, and packet prioritization through wire-speed QoS. (For a true HD voice experience, an HD-compatible phone is necessary. This includes most corded telephones, and the Ooma Telo handset.)
To combat the packet loss that some VoIP users experience as garbled or interrupted voice signals, Ooma Telo’s PureVoice HD also incorporates adaptive redundancy — the Ooma Telo VoIP home phone system detects packet loss and issues duplicate packets to cover the gap. ...
- How does the Ooma Telo phone system work?
- Ooma Telo Calling Plan
- Vocal clarity with Ooma Telo’s PureVoice HD
- What equipment do you need to use the Ooma Telo phone system?
- Keep your phone number
- Use the Ooma phone system anywhere in the world
- Ooma Telo Prices
- Cost Benefit of Ooma Telo Phone System
- Ooma Telo Premier
- Ooma Telo on your iPhone or Android
- See also
The Ooma Telo is a simple, plug-and-play solution for accessing residential VoIP (Voice over Internet Protocol) service.
The Ooma Telo is a VoIP adapter, which enables residential users with a high-speed Internet connection to access inexpensive VoIP phone services. With Ooma Telo VoIP, residential Internet customers have access to:
- Free long-distance calls
- International calls as low as 1¢ per minute
- Free caller ID
- Free call waiting
- Free voicemail.
The Ooma Telo VoIP home phone system also offers extension service with Ooma Telo softphone apps for Android and iPhone mobile phone users.
How does the Ooma Telo phone system work?
You might think of the Ooma Telo as a base, much like a cable modem. The Ooma Telo is an analog telephone adapter (ATA) that connects to an Ethernet cable. You then connect your telephone to the Ooma Telo VoIP adapter. You do not have to connect your computer to the Ooma Telo VoIP home phone system.
The Ooma Telo VoIP home phone system functions as more than an ATA because of included telephony features. In addition to converting your analog phone signal to a digital phone signal, the Ooma Telo phone system includes free inbound and outbound caller ID, call waiting, and voicemail. Ooma Telo Premier subscribers enjoy the benefits of a second line, 3-way conference calling, voicemail-to-text transcription, and more.
Ooma Telo Calling PlanFor international calls, the Ooma Telo phone system offers two options: standard rate and plan rate. With Ooma Telo’s standard rate option, there is no monthly fee or commitment, and calls to 61 countries are usually lower than 1¢ per minute. The Ooma Telo plan rate offers 1,000 minutes of international calling for under $10 per month.
There are a few exceptions to free local and long distance calls in the United States. With the Ooma Telo VoIP home phone system, 411 calls will cost .99¢ and be charged to a prepaid phone card. Also, 900 numbers and other phone chat service numbers are not supported.
Vocal clarity with Ooma Telo’s PureVoice HD
The Ooma Telo VoIP home phone system also claims to offer better clarity than Ooma Telo VoIP competitors with its PureVoice HD technology. The Ooma Telo uses HD voice (double the voice signal), encrypted calls, advanced voice compression, and packet prioritization through wire-speed QoS. (For a true HD voice experience, an HD-compatible phone is necessary. This includes most corded telephones, and the Ooma Telo handset.)
To combat the packet loss that some VoIP users experience as garbled or interrupted voice signals, Ooma Telo’s PureVoice HD also incorporates adaptive redundancy — the Ooma Telo VoIP home phone system detects packet loss and issues duplicate packets to cover the gap. ...
FAXAGE
FAXAGE
The FAXAGE Internet Fax service offers an API integration method that can allow VoIP service providers to provide their own branded electronic fax service to their customers, as an alternative to directly integrating fax capabilities into the service provider's network.
Key features include:
More information is available at our web site: http://www.faxage.com
Direct contact information:
Email - sales@faxage.com
Phone - 303-991-6020 x 200
Toll-Free - 800-853-3293 x 200
The FAXAGE Internet Fax service offers an API integration method that can allow VoIP service providers to provide their own branded electronic fax service to their customers, as an alternative to directly integrating fax capabilities into the service provider's network.
Key features include:
- Choice of HTTPS Post-based API or Email-based interfaces for sending/receiving faxes
- All TDM network designed specifically to support faxing
- SuperG3 (v.34, 33600) fax support
- Hyperfine, fine and low/normal resolutions supported (fine default)
- Supports inbound PDF or TIFF format (PDF default)
- Supports outbound PDF, PostScript, PCL, OpenOffice.org Writer and Spreadsheet, Word doc and docx, Excel xls and xlsx, JPEG, BMP, GIF, TIFF, HTML, Rich text RTF, and CSV
- Local fax DIDs provided in most markets in the US (6,200+ rate centers, 115,000+ exchanges, all 48 states and D.C.)
- Toll-free fax DIDs provided supporting the 50 United States and all of Canada
- Outgoing fax support to the 50 United States and all of Canada
- LNP available for all supported rate centers - website tool to determine portability and create LOA's
- Toll-Free Resporg/transfer for fax numbers available
- Per-channel/per-DID/per-minute pricing structure on an individual case basis
More information is available at our web site: http://www.faxage.com
Direct contact information:
Email - sales@faxage.com
Phone - 303-991-6020 x 200
Toll-Free - 800-853-3293 x 200
Asterisk cmd GotoIfTime
Synopsis:Conditional goto on current time
Description:GotoIfTime(<time range>,<days of week>,<days of month>,<months>?[labeliftrue][:labeliffalse])
If the current time matches the specified time, then branch to the specified extension. Each of the elements may be specified either as '*' (for always) or as a range. If the current time does not match the specified time, and no false target is defined, the next priority is executed.
Times before Asterisk 1.6.2 are only accurate down to the 2-minute interval. So 12:01 is treated the same as 12:00.
Starting with 1.6.2, times are accurate down to the minute.
How to specify timeThe include syntax is defined in the sample extensions.conf like this:
<time range>,<days of week>,<days of month>,<months>
where:
<time range>= <hour>':'<minute>'-'<hour>':'<minute>
| "*"
<days of week> = <dayname>
| <dayname>'-'<dayname>
| "*"
<dayname> = "sun" | "mon" | "tue" | "wed" | "thu" | "fri" | "sat"
<days of month> = <daynum>
| <daynum>'-'<daynum>
| "*"
<daynum> = a number, 1 to 31, inclusive
<hour> = a number, 0 to 23, inclusive
<minute> = a number, 0 to 59, inclusive
<months> = <monthname>
| <monthname>'-'<monthname>
| "*"
<monthname> = "jan" | "feb" | "mar" | "apr" | "may" | "jun" | "jul" | "aug" | "sep" | "oct" | "nov" | "dec"
daynames and monthnames are not case-sensitive.
ExamplesIf you replace an option with *, it is ignored when matching. For instance:
exten => 3000,1,GotoIfTime(9:00-17:00,mon-fri,*,*?open)
would transfer to priority label "open" in the current extension if it's between 9:00 and 17:00, Monday through Friday, not checking the day of month or month.
Another example:
exten => s,n,GotoIfTime(*,*,26-30,May?attendant)
would transfer to priority label "attendant" in the current extension at any time from May 26th though May 30th. (In this example, an office is closed for Memorial Day.)
Holidays
If the comment about holidays is true, then here's a list suitable in the United States:
Independence Day: *,*,4,jul
Christmas: *,*,25,dec
NewYear: *,*,1,jan
MartinLutherKing: *,mon,15-21,jan
Valentines: *,*,14,feb
StPatDay *,*,17,mar
Halloween *,*,31,oct
Thanksgiving *,thu,22-28,nov
MemorialDay *,mon,25-31,may
LaborDay *,mon,1-7,sep
Pres/WashBday *,mon,15-21,feb
MothersDay *,sun,8-14,may
FathersDay *,sun,15-21,jun
Easter: Good Luck! Paschal moons, etc). ...
Description:GotoIfTime(<time range>,<days of week>,<days of month>,<months>?[labeliftrue][:labeliffalse])
If the current time matches the specified time, then branch to the specified extension. Each of the elements may be specified either as '*' (for always) or as a range. If the current time does not match the specified time, and no false target is defined, the next priority is executed.
Times before Asterisk 1.6.2 are only accurate down to the 2-minute interval. So 12:01 is treated the same as 12:00.
Starting with 1.6.2, times are accurate down to the minute.
How to specify timeThe include syntax is defined in the sample extensions.conf like this:
<time range>,<days of week>,<days of month>,<months>
where:
<time range>= <hour>':'<minute>'-'<hour>':'<minute>
| "*"
<days of week> = <dayname>
| <dayname>'-'<dayname>
| "*"
<dayname> = "sun" | "mon" | "tue" | "wed" | "thu" | "fri" | "sat"
<days of month> = <daynum>
| <daynum>'-'<daynum>
| "*"
<daynum> = a number, 1 to 31, inclusive
<hour> = a number, 0 to 23, inclusive
<minute> = a number, 0 to 59, inclusive
<months> = <monthname>
| <monthname>'-'<monthname>
| "*"
<monthname> = "jan" | "feb" | "mar" | "apr" | "may" | "jun" | "jul" | "aug" | "sep" | "oct" | "nov" | "dec"
daynames and monthnames are not case-sensitive.
ExamplesIf you replace an option with *, it is ignored when matching. For instance:
exten => 3000,1,GotoIfTime(9:00-17:00,mon-fri,*,*?open)
would transfer to priority label "open" in the current extension if it's between 9:00 and 17:00, Monday through Friday, not checking the day of month or month.
Another example:
exten => s,n,GotoIfTime(*,*,26-30,May?attendant)
would transfer to priority label "attendant" in the current extension at any time from May 26th though May 30th. (In this example, an office is closed for Memorial Day.)
Holidays
If the comment about holidays is true, then here's a list suitable in the United States:
Independence Day: *,*,4,jul
Christmas: *,*,25,dec
NewYear: *,*,1,jan
MartinLutherKing: *,mon,15-21,jan
Valentines: *,*,14,feb
StPatDay *,*,17,mar
Halloween *,*,31,oct
Thanksgiving *,thu,22-28,nov
MemorialDay *,mon,25-31,may
LaborDay *,mon,1-7,sep
Pres/WashBday *,mon,15-21,feb
MothersDay *,sun,8-14,may
FathersDay *,sun,15-21,jun
Easter: Good Luck! Paschal moons, etc). ...
Asterisk queue_log on MySQL
Asterisk 1.8The table format and behavior of realtime queue_log storage has changed for Asterisk 1.8.
More details: https://issues.asterisk.org/view.php?id=17082
SVN Trunk and Asterisk 1.6.x
Current SVN trunk (Revision 94782 from 12-26-07 09:54), supports storage of queue_log in your RT engine.
More details: http://bugs.digium.com/view.php?id=11625
PLEASE NOTE: The table structure referenced in the above link is no longer accurate (use the structure below). The time column must be char(10), or you will receive an error like this:
WARNING[25801] res_config_mysql.c: Realtime table general@queue_log: column 'time' cannot be type 'int(10) unsigned' (need char)
Sample queue_log table for MySQL:
CREATE TABLE `queue_log` ( `id` int(10) unsigned NOT NULL auto_increment, `time` char(10) unsigned default NULL, `callid` varchar(32) NOT NULL default '', `queuename` varchar(32) NOT NULL default '', `agent` varchar(32) NOT NULL default '', `event` varchar(32) NOT NULL default '', `data` varchar(255) NOT NULL default '', PRIMARY KEY (`id`) );
queue_log message consists of 2 parts: constant and variable. Constant part is split among corresponding fields (time, callid, queuename, agent, event). Variable data is stored in `data` field as is, so you'll meet '|' there. Example:
mysql> select * from queue_log; +----+------------+--------------+------------------+-------+------------+-------+ | id | time | callid | queuename | agent | event | data | +----+------------+--------------+------------------+-------+------------+-------+ | 1 | 1198356717 | 1198356717.0 | voipsolutions.ru | NONE | ENTERQUEUE | |serg | | 2 | 1198356719 | 1198356717.0 | voipsolutions.ru | NONE | ABANDON | 1|1|2 | +----+------------+--------------+------------------+-------+------------+-------+
to activate RT logging add a line like
queue_log => mysql,asterisk
to your extconfig.conf
PLEASE NOTE: Asterisk 1.6 changed the way that the extconfig.conf file references the MySQL database. You now specify the context name in extconfig.conf, NOT the database name.
Converting to a more readable formatSo, you've got queue_log created from realtime, but it doesn't make easy parsing. Here's a new table definition and some SQL code to pretty it up for you. Note that locking etc. aren't done - we run this job at night when we know there aren't any calls anyway, but your situation may differ.
CREATE TABLE IF NOT EXISTS `queue_log_processed` (
`recid` int(10) unsigned NOT NULL auto_increment,
`origid` int(10) unsigned NOT NULL,
`callid` varchar(32) NOT NULL default '',
`queuename` varchar(32) NOT NULL default '',
`agentdev` varchar(32) NOT NULL,
`event` varchar(32) NOT NULL default '',
`data1` varchar(128) NOT NULL,
`data2` varchar(128) NOT NULL,
`data3` varchar(128) NOT NULL,
`datetime` datetime NOT NULL default '0000-00-00 00:00:00',
PRIMARY KEY (`recid`),
KEY `data1` (`data1`),
KEY `data2` (`data2`),
KEY `data3` (`data3`),
KEY `event` (`event`),
KEY `queuename` (`queuename`),
KEY `callid` (`callid`),
KEY `datetime` (`datetime`),
KEY `agentdev` (`agentdev`),
KEY `origid` (`origid`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=5 ;
And then we run the following SQL:
INSERT INTO queue_log_processed (origid,callid,queuename,agentdev,event,data1,data2,data3,datetime)
SELECT id,callid,queue_log. ...
More details: https://issues.asterisk.org/view.php?id=17082
SVN Trunk and Asterisk 1.6.x
Current SVN trunk (Revision 94782 from 12-26-07 09:54), supports storage of queue_log in your RT engine.
More details: http://bugs.digium.com/view.php?id=11625
PLEASE NOTE: The table structure referenced in the above link is no longer accurate (use the structure below). The time column must be char(10), or you will receive an error like this:
WARNING[25801] res_config_mysql.c: Realtime table general@queue_log: column 'time' cannot be type 'int(10) unsigned' (need char)
Sample queue_log table for MySQL:
CREATE TABLE `queue_log` ( `id` int(10) unsigned NOT NULL auto_increment, `time` char(10) unsigned default NULL, `callid` varchar(32) NOT NULL default '', `queuename` varchar(32) NOT NULL default '', `agent` varchar(32) NOT NULL default '', `event` varchar(32) NOT NULL default '', `data` varchar(255) NOT NULL default '', PRIMARY KEY (`id`) );
queue_log message consists of 2 parts: constant and variable. Constant part is split among corresponding fields (time, callid, queuename, agent, event). Variable data is stored in `data` field as is, so you'll meet '|' there. Example:
mysql> select * from queue_log; +----+------------+--------------+------------------+-------+------------+-------+ | id | time | callid | queuename | agent | event | data | +----+------------+--------------+------------------+-------+------------+-------+ | 1 | 1198356717 | 1198356717.0 | voipsolutions.ru | NONE | ENTERQUEUE | |serg | | 2 | 1198356719 | 1198356717.0 | voipsolutions.ru | NONE | ABANDON | 1|1|2 | +----+------------+--------------+------------------+-------+------------+-------+
to activate RT logging add a line like
queue_log => mysql,asterisk
to your extconfig.conf
PLEASE NOTE: Asterisk 1.6 changed the way that the extconfig.conf file references the MySQL database. You now specify the context name in extconfig.conf, NOT the database name.
Converting to a more readable formatSo, you've got queue_log created from realtime, but it doesn't make easy parsing. Here's a new table definition and some SQL code to pretty it up for you. Note that locking etc. aren't done - we run this job at night when we know there aren't any calls anyway, but your situation may differ.
CREATE TABLE IF NOT EXISTS `queue_log_processed` (
`recid` int(10) unsigned NOT NULL auto_increment,
`origid` int(10) unsigned NOT NULL,
`callid` varchar(32) NOT NULL default '',
`queuename` varchar(32) NOT NULL default '',
`agentdev` varchar(32) NOT NULL,
`event` varchar(32) NOT NULL default '',
`data1` varchar(128) NOT NULL,
`data2` varchar(128) NOT NULL,
`data3` varchar(128) NOT NULL,
`datetime` datetime NOT NULL default '0000-00-00 00:00:00',
PRIMARY KEY (`recid`),
KEY `data1` (`data1`),
KEY `data2` (`data2`),
KEY `data3` (`data3`),
KEY `event` (`event`),
KEY `queuename` (`queuename`),
KEY `callid` (`callid`),
KEY `datetime` (`datetime`),
KEY `agentdev` (`agentdev`),
KEY `origid` (`origid`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=5 ;
And then we run the following SQL:
INSERT INTO queue_log_processed (origid,callid,queuename,agentdev,event,data1,data2,data3,datetime)
SELECT id,callid,queue_log. ...
Asterisk system vendors
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- Africa
- Asia and the Pacific
- Australia
- Bangladesh
- Bahrain
- China
- Hong Kong
- India
- Indonesia
- Japan
- Kingdom of Saudi Arabia
- Malaysia
- New Zealand
- Pakistan
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- Uncategorized
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- Austria
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- Sweden
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- UK
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- Middle East
- UAE, Oman, Bahrain, Qatar, Saudi Arabia, Kuwait
- UAE, Oman, Bahrain, Qatar, Saudi Arabia, Kuwait
- Egypt
- Iran
- Israel
- Latin America
- Bolivia
- Colombia
- Costa Rica
Hosted VoIP
Hosted VoIP is another term for Hosted PBX. "Hosted" means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP providers data center, thus the term: hosted VoIP.
Benefits of Hosted VoIP
There are many benefits to using Hosted VoIP rather than a traditional phone system, or an on-premise VoIP system. The main benefit is cost- a Hosted VoIP system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted VoIP system. Hosted VoIP phone systems fall under operational expenditure rather than capital expenditure, which also make hosted VoIP systems attractive to businesses.
Another benefit to a hosted VoIP system over an on-premise PBX is that hosted VoIP providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get hosted VoIP.
Hosted VoIP Service Providers
Please list hosted VoIP service providers below in alphabetical order.
VMStorm VPS provides reliable VOIP Hosting in Pennsylvania.
ZONE Limited launches hosted PBX service where you can:
Please contact sales@zonetel.com for more details.
See also
Benefits of Hosted VoIP
There are many benefits to using Hosted VoIP rather than a traditional phone system, or an on-premise VoIP system. The main benefit is cost- a Hosted VoIP system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted VoIP system. Hosted VoIP phone systems fall under operational expenditure rather than capital expenditure, which also make hosted VoIP systems attractive to businesses.
Another benefit to a hosted VoIP system over an on-premise PBX is that hosted VoIP providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get hosted VoIP.
Hosted VoIP Service Providers
Please list hosted VoIP service providers below in alphabetical order.
VMStorm VPS provides reliable VOIP Hosting in Pennsylvania.
ZONE Limited launches hosted PBX service where you can:
- Enjoy a full set of advanced PBX features without upfront hardware investment
- Pay as you go
- Have customized IVR and greetings
- Subscribe China, Hong Kong and Singapore DID numbers to gain local presence in these regions
- Use SIP phones or mobiles (PSTN forward) to answer calls
- A-Z international termination at attractive rates
Please contact sales@zonetel.com for more details.
See also
- Managed PBX
- VOIP Service Providers
- VOIP Service Providers Business
Hosted PABX
Hosted PABX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.
.e4 Technologies - .e4 provides powerful, integrated IP Communications enterprise class services that provide the SMB customer to configure the exact voice, video, messaging and collaboration services they need in just minutes- all delivered on-demand.
CALL 877-434-VoIP
Not like any other hosted IP pbx service provider.
What's different?
Of course, free trial!
Give us a call at 1-800-801-3381 or visit us at www.onsip.com.
Ip-Pabx.com IP-Pabx.com - First of its kind Hosted IP Based Pbx system offering from A-Z Solution from one provider. Totally Hosted Web Based Pbx System, Very Low priced starting 199$. You place an order, choose your Country Code and Area code for your incoming lines, choose extension quantity, They ship you pre programmed 99$ ip phones, you plug them on the internet, and wala, you have a pbx system up and running. the extensions can be anywhere in the world, and they will work fine. visit the web site for demo, or email us for a free test account. You get Voice Mail, Fax to Email, Web Voice Mail, Call Recording, 20 IVR's. API to integrate your own applications.
Sonetel A free, global phone system for small and medium sized companies worldwide. This is a free IP-PABX service that can be connected to any phone company. ...
.e4 Technologies - .e4 provides powerful, integrated IP Communications enterprise class services that provide the SMB customer to configure the exact voice, video, messaging and collaboration services they need in just minutes- all delivered on-demand.
CALL 877-434-VoIP
- 1 MONTH FREE TRIAL
- VoIP and PSTN Service
- PBX and ACD
- Web, Audio & Video Conferencing
- Skills Based Routing
- Private/Encrypted and Public IM
- Voice and Data Archiving
- Reporting and Archiving
- PSTN Gateway
- SIP Hardware Compatibility
- CRM Integration
- Web Contact Center
Not like any other hosted IP pbx service provider.
What's different?
- We charge for usage, NOT users. You'll end up paying less, A LOT less. Why pay for conference room phones?
- You can have TEN SIP phones registered for the same user. That means home phone, mobile SIP client and your desk phone all ring how and when you want them to!
- Use ANY SIP phones. SIP is in our name for a reason. We review the most popular IP phones to help you choose the right ones.
- No other provider has a private, standards based Instant Messageing service in a browser with click-to-call, phone presence and more.
- Plenty of Extras built by us and others using our open API.
Of course, free trial!
Give us a call at 1-800-801-3381 or visit us at www.onsip.com.
Ip-Pabx.com IP-Pabx.com - First of its kind Hosted IP Based Pbx system offering from A-Z Solution from one provider. Totally Hosted Web Based Pbx System, Very Low priced starting 199$. You place an order, choose your Country Code and Area code for your incoming lines, choose extension quantity, They ship you pre programmed 99$ ip phones, you plug them on the internet, and wala, you have a pbx system up and running. the extensions can be anywhere in the world, and they will work fine. visit the web site for demo, or email us for a free test account. You get Voice Mail, Fax to Email, Web Voice Mail, Call Recording, 20 IVR's. API to integrate your own applications.
Sonetel A free, global phone system for small and medium sized companies worldwide. This is a free IP-PABX service that can be connected to any phone company. ...
ZONE Limited
http://www.zonetel.com
Established in 2000, ZONE Limited is a licensed telecom service and solution provider in Hong Kong. ZONE specializes in VOIP/SIP/Asterisk PBX/Call Center/IVR and have a number of successful implementations in Asian regions. The management team strives to provide quality service and achieve customer satisfaction.
NEWTelephony and PBX Cloud ServiceHost your IP telephony applications in our Asterisk cloud environment in Asia regions: China, Hong Kong and Singapore.
Please visit Cloud Plans for price quotes.
ServicesDID numbers/Hong Kong Virtual Phone NumberZONE is a licensed telecom operator in HK offering HK DID numbers (HK virtual phone numbers) to your compatible SIP platforms like Asterisk, Elastix and many others. You can get HK local presence without all the expensive overhead. The DID comes with optional outbound dialing, voice recording, voice mail, call forward and concurrent call features.
The DID is applicable in call centers and virtual offices.
In addition to HK DID, we provide Singapore and China DID for you to develop the Asia markets.
IP-PBXWe carry IP-PBX products from selected brands which we believe to represent the best mix to serve SME/SOHO segments. We provide on-site installation, integration, customization and training so that customers can make full use of the IP technology.
Asterisk consultation and turnkey solutionsAsterisk is known to be the leading open-source PBX attributed to its powerful features and deployment flexibility. We have intensive exposure to Asterisk and could help you to implement various business applications such as calling card, IVR, voice recording, fax2email, auto-attendant, dialers, etc.
Hosted PBX and IVR serviceLooking for a hosted PBX service? We provide elastix/asterisk PBX hosting environment with DID SIP trunks and support of remote extensions. Pay-as-go-you.
Or need a hosted IVR service to integrate with your business applications ? We are able to deliver advanced features like call queues with customizable greetings, operator-assisted key, fax-on-demand, time-dependent IVR options, database integration and many.
SMS service with international coverageIt is convenient to broadcast and schedule SMS on our web site. In addition, we act as a programmable SMS gateway supporting the standard smpp and http interface.
A-Z SIP terminationZONE has been operating IDD business in HK for more than 10 years. We partner with many international carriers to provide a reliable and quality voice service. We have also extended the IDD service via VOIP/SIP so that you could enjoy our attractive A-Z rates from anywhere in the globe.
Web callbackLaunching toll-free access to your customer service center becomes easy and cost-effective with our web callback. It just involves a few lines of codes in your web server to enable the toll-free channel with every usage statistics. ...
Established in 2000, ZONE Limited is a licensed telecom service and solution provider in Hong Kong. ZONE specializes in VOIP/SIP/Asterisk PBX/Call Center/IVR and have a number of successful implementations in Asian regions. The management team strives to provide quality service and achieve customer satisfaction.
NEWTelephony and PBX Cloud ServiceHost your IP telephony applications in our Asterisk cloud environment in Asia regions: China, Hong Kong and Singapore.
- Offer your customer the optimized access speed.
- Pay-as-you-go.
- Bundled with suite of value added services like hosted PBX, SIP trunking and DID numbers.
Please visit Cloud Plans for price quotes.
ServicesDID numbers/Hong Kong Virtual Phone NumberZONE is a licensed telecom operator in HK offering HK DID numbers (HK virtual phone numbers) to your compatible SIP platforms like Asterisk, Elastix and many others. You can get HK local presence without all the expensive overhead. The DID comes with optional outbound dialing, voice recording, voice mail, call forward and concurrent call features.
The DID is applicable in call centers and virtual offices.
In addition to HK DID, we provide Singapore and China DID for you to develop the Asia markets.
IP-PBXWe carry IP-PBX products from selected brands which we believe to represent the best mix to serve SME/SOHO segments. We provide on-site installation, integration, customization and training so that customers can make full use of the IP technology.
Asterisk consultation and turnkey solutionsAsterisk is known to be the leading open-source PBX attributed to its powerful features and deployment flexibility. We have intensive exposure to Asterisk and could help you to implement various business applications such as calling card, IVR, voice recording, fax2email, auto-attendant, dialers, etc.
Hosted PBX and IVR serviceLooking for a hosted PBX service? We provide elastix/asterisk PBX hosting environment with DID SIP trunks and support of remote extensions. Pay-as-go-you.
Or need a hosted IVR service to integrate with your business applications ? We are able to deliver advanced features like call queues with customizable greetings, operator-assisted key, fax-on-demand, time-dependent IVR options, database integration and many.
SMS service with international coverageIt is convenient to broadcast and schedule SMS on our web site. In addition, we act as a programmable SMS gateway supporting the standard smpp and http interface.
A-Z SIP terminationZONE has been operating IDD business in HK for more than 10 years. We partner with many international carriers to provide a reliable and quality voice service. We have also extended the IDD service via VOIP/SIP so that you could enjoy our attractive A-Z rates from anywhere in the globe.
Web callbackLaunching toll-free access to your customer service center becomes easy and cost-effective with our web callback. It just involves a few lines of codes in your web server to enable the toll-free channel with every usage statistics. ...
Asterisk tips openhours
How to include contexts based on time and dateThe Asterisk dial plan, extensions.conf lets you include contexts based on time and/or date.
Syntax:include => <context>,<times>,<weekdays>,<mdays>,<months>
Example:; First, let's do the holidays.
include => holiday,*,*,1,jan
include => holiday,*,*,31,may
include => holiday,*,*,4,jul
include => holiday,*,*,6,sep
include => holiday,17:00-23:59,*,24,nov
include => holiday,*,*,25,nov
include => holiday,17:00-23:59,*,24,dec
include => holiday,*,*,25,dec
include => holiday,17:00-23:59,*,31,dec
; These are the days we're open.
include => day,09:00-19:59,mon-fri,*,*
include => day,10:00-14:59,sat,*,*
; If we're not open, we're closed.
include => night
Example from Troy Settle
I ran into a problem with this. The night context is always included in this scenario, and if the night context contains the same extensions as the day context, Asterisk will continue with the night context if the extensions match:
[day]
exten => s,8,BackGround(to-hear-menu-again)
will jump to:
[night]
exten => s,9,BackGround()
To prevent this, specify an explicit night include:
include => night,20:00-8:59,mon-fri,*,*
include => night,15:00-9:59,sat,*,*
include => night,*,sun,*,*
Syntax:include => <context>,<times>,<weekdays>,<mdays>,<months>
Example:; First, let's do the holidays.
include => holiday,*,*,1,jan
include => holiday,*,*,31,may
include => holiday,*,*,4,jul
include => holiday,*,*,6,sep
include => holiday,17:00-23:59,*,24,nov
include => holiday,*,*,25,nov
include => holiday,17:00-23:59,*,24,dec
include => holiday,*,*,25,dec
include => holiday,17:00-23:59,*,31,dec
; These are the days we're open.
include => day,09:00-19:59,mon-fri,*,*
include => day,10:00-14:59,sat,*,*
; If we're not open, we're closed.
include => night
Example from Troy Settle
- Asterisk config extensions.conf: The dial plan
- Asterisk cmd gotoiftime: Conditional goto based on time/date
I ran into a problem with this. The night context is always included in this scenario, and if the night context contains the same extensions as the day context, Asterisk will continue with the night context if the extensions match:
[day]
exten => s,8,BackGround(to-hear-menu-again)
will jump to:
[night]
exten => s,9,BackGround()
To prevent this, specify an explicit night include:
include => night,20:00-8:59,mon-fri,*,*
include => night,15:00-9:59,sat,*,*
include => night,*,sun,*,*
Fake False Answer Supervision (FAS) service
Warning!
The service described below is what most would consider fraud and is illegal in many countries.
The information is presented only to assist in identifying this type of fraud.
Fake FAS service is a service that allows you to earn extra money by mixing fake FAS calls into your real traffic. The service generates the needed carrier service messages and creates the needed billable airtime of your traffic, thus, simulating calls to numbers that are "out of coverage" and providing false billing.
WHAT IS FAS?
FAS is commonly referred to as False Answer Supervision and for a regular user this phenomenon means incorrect extra billing of calls: the billing starts earlier than the called party actually picks up the phone.
What is Fake FAS service?
Fake FAS service is a service that simulates calls to numbers that are out of network coverage and that provides false billable airtime to the calling party. The service pretends to be a real mobile carrier and plays back real mobile carrier service messages (e.g. "the number you are calling is not reachable at the moment, please, call back later") while charging for this.
HOW TO IDENTIFY FAS?
Once you notice one of the following, you are sure to experience a FAS problem :
- the billing of the call starts earlier than the called party picks up the phone
- the call is billed even if the called party is said to be out of the coverage (for example, for a mobile phone you get an announcement "we are sorry, but the party you are calling is unavailable at the moment")
- the call is billed even if the caller gets to a voicemail.
WHY DO WE EXPERIENCE FAS?
FAS normally takes place when there is no synchronization between a VoIP leg and PSTN leg of the call on a VoIP-to-PSTN gateway. When the call reaches the gateway from the VoIP network, the gateway tries to establish a connection with the called number, but due to incorrect configuration it cannot detect the states of the call, which are advertised by the PSTN network (the states are: "called party ringing", "called party connected"). And thus the gateway forces the "CONNECT" state and it normally happens immediately after the arrival of the call from the VoIP network or a few seconds after that. But the main idea is that the gateway connects the calls (which means starts the part of the call, which is billed) according to its own settings, but not according to the actual state of the call.
WHY FAS IS A "BAD THING"?
FAS is considered to be a negative phenomenon due to two reasons:
1. The calling enduser may notice that he/she is billed for non-connected calls
2. FAS creates a bigger number of short duration calls (in a situation when there is no answer from the called party, the connected duration of the call is very short (as long as the calling party is willing to listen to Ring Back Tones without being answered), which brings down the ACD/ALOC (average call duration / average length of call) (due to a bigger than usual number of short calls) and brings up the ASR (average success rate) (due to a bigger than usual number of connected calls).
HOW TO EARN MONEY USING FAS?
Normally this is a prerogative of the owners of the gateways that connect the VoIP network to PSTN network. The extra penny is earned by charging the calling party a bit more for the calls using the ability to control the "CONNECT" state of the call. But wholesale transit carriers can also take advantage of using FAS phenomenon to increase profit: by using our service it is possible to mix the needed percent of FAS calls into your existing traffic. This will slightly degrade the ACD parameter of your traffic, increase the ASR parameter and will allow you to earn extra money out of almost nowhere.
HOW DOES IT WORK?
You put the FAS server into routing on your switch. The calls that come to the FAS server will be forwarded to an IVR message, related to the dialed number. The message that will be played back will sound something like this: "the subscriber you are calling is out of reach, please, try to call later". (IMPORTANT NOTE: it is your responsibility to record such a voice message and to upload it to FAS server.) The message should be exactly the same as the caller would get from the real carrier, to whom this dialed number belongs. The call will be billed from the very beginning and this will be the billable time that will generate additional profit. ...
The service described below is what most would consider fraud and is illegal in many countries.
The information is presented only to assist in identifying this type of fraud.
Fake FAS service is a service that allows you to earn extra money by mixing fake FAS calls into your real traffic. The service generates the needed carrier service messages and creates the needed billable airtime of your traffic, thus, simulating calls to numbers that are "out of coverage" and providing false billing.
WHAT IS FAS?
FAS is commonly referred to as False Answer Supervision and for a regular user this phenomenon means incorrect extra billing of calls: the billing starts earlier than the called party actually picks up the phone.
What is Fake FAS service?
Fake FAS service is a service that simulates calls to numbers that are out of network coverage and that provides false billable airtime to the calling party. The service pretends to be a real mobile carrier and plays back real mobile carrier service messages (e.g. "the number you are calling is not reachable at the moment, please, call back later") while charging for this.
HOW TO IDENTIFY FAS?
Once you notice one of the following, you are sure to experience a FAS problem :
- the billing of the call starts earlier than the called party picks up the phone
- the call is billed even if the called party is said to be out of the coverage (for example, for a mobile phone you get an announcement "we are sorry, but the party you are calling is unavailable at the moment")
- the call is billed even if the caller gets to a voicemail.
WHY DO WE EXPERIENCE FAS?
FAS normally takes place when there is no synchronization between a VoIP leg and PSTN leg of the call on a VoIP-to-PSTN gateway. When the call reaches the gateway from the VoIP network, the gateway tries to establish a connection with the called number, but due to incorrect configuration it cannot detect the states of the call, which are advertised by the PSTN network (the states are: "called party ringing", "called party connected"). And thus the gateway forces the "CONNECT" state and it normally happens immediately after the arrival of the call from the VoIP network or a few seconds after that. But the main idea is that the gateway connects the calls (which means starts the part of the call, which is billed) according to its own settings, but not according to the actual state of the call.
WHY FAS IS A "BAD THING"?
FAS is considered to be a negative phenomenon due to two reasons:
1. The calling enduser may notice that he/she is billed for non-connected calls
2. FAS creates a bigger number of short duration calls (in a situation when there is no answer from the called party, the connected duration of the call is very short (as long as the calling party is willing to listen to Ring Back Tones without being answered), which brings down the ACD/ALOC (average call duration / average length of call) (due to a bigger than usual number of short calls) and brings up the ASR (average success rate) (due to a bigger than usual number of connected calls).
HOW TO EARN MONEY USING FAS?
Normally this is a prerogative of the owners of the gateways that connect the VoIP network to PSTN network. The extra penny is earned by charging the calling party a bit more for the calls using the ability to control the "CONNECT" state of the call. But wholesale transit carriers can also take advantage of using FAS phenomenon to increase profit: by using our service it is possible to mix the needed percent of FAS calls into your existing traffic. This will slightly degrade the ACD parameter of your traffic, increase the ASR parameter and will allow you to earn extra money out of almost nowhere.
HOW DOES IT WORK?
You put the FAS server into routing on your switch. The calls that come to the FAS server will be forwarded to an IVR message, related to the dialed number. The message that will be played back will sound something like this: "the subscriber you are calling is out of reach, please, try to call later". (IMPORTANT NOTE: it is your responsibility to record such a voice message and to upload it to FAS server.) The message should be exactly the same as the caller would get from the real carrier, to whom this dialed number belongs. The call will be billed from the very beginning and this will be the billable time that will generate additional profit. ...
