| What hardware is compatible with Callcentric? |
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In general most SIP/2.0 hardware should be compatible with Callcentric. As a BYOD (Bring Your Own Device) provider we do not restrict you to using only the hardware we're comfortable with. Users have had success using software from companies such as:
Cisco, Grandstream, Sipura, Linksys, OBiHai, Snom, AVM, DrayTek, Epygi, Polycom, Gigaset, Panasonic, CloudTC, Nokia, iPhone, Android, Avaya, Aastra, Yealink and many more.
Hardware from other vendors surpporting SIP/2.0 should be compatible.
You can view a list of the hardware we have created setup guides for below:
ATA Adaptors
IP Phones
Cordless DECT IP Phones
Mobile Softphones
If you have questions about other hardware please contact us. |
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| What software is compatible with Callcentric? |
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In general most SIP/2.0 software should be compatible with Callcentric. As a BYOD (Bring Your Own Device) provider we do not restrict you to using only the software we're comfortable with. Users have had success using software from companies such as:
3CX, NCH, Digium Asterisk, Counterpath, Acrobits, Snom, Twinkle, freePBX, Android, Nokia Symbian, LinPhone, fring and many other open source and proprietary applications.
You can view a list of the software we have created setup guides for below:
Desktop Softphones
Mobile Softphones
IP PBX Software
If you have questions about other software please contact us. |
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| Can I use my hardware/software with your service? |
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Yes assuming that we support it. Many devices are currently supported and in general you can use any device that supports at least the following:
- SIP 2.0
- RTP over UDP
- DIGEST MD5 authentication
- Fixed Outbound Proxy
- DNS SRV
- G.711 and/or G.729
If after reviewing the supported equipment list you are still unsure if your device will work, please contact us. |
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| How do I setup my telephone adapter? |
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We provide setup guides for the most common UA (hardware or software) used with Callcentric. You can click one of the categories below to view different types of UA and find the guide which would work best for you.
| Category | Description |
|---|
| ATA Adaptors | Such as the Linksys PAP2, Cisco SPA and Grandstream HandyTone | | IP Phones | Such as the Grandstream GXV3140 Video phone, Gigaset DX800A, Polyom Soundpint and snom 820 | | Cordless IP Phones | Such as the Gigaset C610A, A510 and snom m9 | | IP PBX Software | From Asterisk, snom, 3CX and more | | Desktop Softphones | Such as X-Lite, ZoIPer, Jitsi, LinPhone and more | | Mobile Softphones | Such as the popular Acrobits, cSIPSimple and native clients for Nokia and Android phones |
If none of the above match what you are trying to use then you may use the geenric settings that should work for most devices with our Generic / Other setup guide. |
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| What do I need to start using Callcentric? |
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If after looking at our How It Works page you are still unsure what you need to do to actually use our services the please read the following:
*If you are trying to use the Callcentric service from countries which specifically block VoIP/SIP, such as the UAE, then you may not be able to use our services. You may still contact us for further information regarding these specific countries however please read the information below before doing so.
1 - First make sure that you have a connection which is not being blocked by your ISP or network, with the necessary bandwidth that is stable enough for VoIP.
2 - You will also want to make sure that you have selected SIP 2.0 compatible hardware or software. We support many different devices and software so you may choose to use your existing SIP 2.0 software or unlocked hardware with our services.
Once you have done the above simply signup for an account and then follow this guide to assist you with testing our services.
*If after running the Visualware test you receive a message stating that your SIP ports are blocked and you are not in a country known for blocking SIP/VoIP you can easily open a trouble ticket on your account for further help. |
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| What voice codecs does Callcentric support? |
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For calls from one Callcentric user to another any codec can be used.
For calls to the voicemail system and for error prompting the following codec's can be used:
G.711 u-law
G.711 A-law
G.729A
For calls to the PSTN (traditional landline and mobile phones) the following codec's can be used:
G.711 u-law
G.711 A-law
G.729A
For calls from the PSTN to your Real Phone Number the following codec's will be used:
G.711 u-law
G.729A
We recommend that you configure your user agent's codec selection in the following order (with 1 being the highest priority, and 3 the lowest), assuming your user agent supports each of the following. If your user agent doesn't support all of these codec's, just skip the ones that it does not have.
1 - G.711 u-law
2 - G.711 A-law
3 - G.729
All other codecs should be disabled for compatibility reasons. If you do enable codecs which are not listed above then keep in mind that this will only work with calls on the Callcentric network or to SIP URIs. If you wish to configure other codecs then the following order may be considered:
4 - G.722 48Kbps
5 - G.722 56Kbps
6 - G.722 64Kbps
7 - Speex 8Kbps
8 - Speex 16Kbps
9 - Speex 32Kbps
10 - G.726 16Kbps
11 - G.726 24Kbps
12 - G.726 32Kbps
13 - G.728
14 - iLBC |
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| What can I manage by logging into my account? |
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Here's a brief list of the things you can manage by logging into your account:
Contact Information
Name, Address, Telephone #, email addresses
Login/General information
Username and password for web access
Username and password for SIP device/software access
Forgotten password question and answer
Time zone
Calling Preferences
Caller ID to send
Do not disturb
Caller ID blocking
Call waiting
Anonymous call rejection
Voicemail Preferences
Voice mail timeout
Voice mail email alerts
Phonebook/directory
Phone book / Speed dial
Reports
Tracking of all inbound and outbound calls and voice mail received
View transactions
View bill statements
View call history
Products and Services
Rate plans ordering and modifications
Voice Mail ordering and modifications
Real phone number/DID ordering and modifications
Billing Settings
Auto recharge settings
Add funds to your balance
Add and remove credit cards
Rates
Download our rates in CSV format by clicking on the Rates link at the top of your My Callcentric account. |
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| What credentials do I need to register to the Callcentric servers? |
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In order to register your UA (User Agent) to the Callcentric servers you will need the correct login information. Specifically you will need the following:
| Username/Auth ID: | Your assigned 1777 number | | Password: | The password you created initially. You may also change your SIP/Phone password here | | Proxy Server: | sip.callcentric.net | | Domain/Realm: | sip.callcentric.net | | Registrar Server: | sip.callcentric.net | | Port: | 5060 |
This is the general information you will need to configure your UA with. You can find more detailed information by visiting our support guide page here, or if your UA is not listed you may follow our "Generic / Other" guide here. |
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| How can I recover my SIP / Phone password? |
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Your Phone/SIP Password is configurable per extension. You will need to use the correct password for the extension you are trying to configure.
If you forgot your SIP / Phone password, used to configure your UA for an extension, then you can follow the instructions below:
- Login to your My Callcentric account
- Click on the EXTENSIONS link in the navigation menu
- Click the Modify link for the extension you wish to edit
- Modify the password for your chosen extension
All changes take effect immediately. You will need to update your UA with the new password in order for registrations to continue working for the extension you modified. |
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| I've changed my SIP/Phone password numerous times but can't register my UA... What's wrong? |
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If you have already changed your SIP/Phone password and are still having problems registering to your Callcentric account due to your password not taking effect then please follow the instructions below:
1. Login to your UA (User Agent), software/hardware, and update it with the new password you wish to use
2. Once you have done this save your changes and either unplug the hardware or exit the software to completely disable it at this time
3. Modify your extension password using the instructions here
At this point your UA should properly register to your Callcentric account.
The reason for this is because login information is cached in the Callcentric authentication servers. You need to first make sure that your UA stops attempting to use your old login information, which would cause that information to be updated and may increase the time it takes to change your login information. Keep in mind that this is not the same for all UA however these instructions should work for most scenarios where password changes do not take immediate effect. |
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| How can another SIP user not using Callcentric call my Callcentric phone? |
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If someone you know has a SIP phone and would like to call your Callcentric phone, the easiest way is to get them to SIGN UP for a FREE Callcentric account.
If however they would like to dial you directly from their current phone with another provider, this can be done a few ways:
1. You can be reached at: [email protected] or [email protected]. A specific example would be:
[email protected]
OR
[email protected]
*Where MYCCNUM is your full DID.
2. Any person/service which peers with SIP Broker can call a Callcentric user by dialing the Callcentric SIP Code - *462 - and the users Callcentric number (1777xxxxxxx). An example would be:
*4621777MYCCID
Please note that when dialing from another network you may have to dial a SIP Broker access code, then the SIP code for Callcentric and the users Callcentric number. For example from FWD you would dial: **275*46217770000001 (**275 is to access SIP Broker from FWD, *462 is the SIP Code for Callcentric on SIP Broker, and 17770000001 is a test number at Callcentric). |
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| Why can't I receive calls from outside Callcentric when using a Callcentric number as the callerID? |
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Calls are not routed solely based on caller ID; however the caller ID is taken in to account for some routing and billing purposes. Because of this, on an incoming call from outside the Callcentric network, if you pass a caller ID of a Callcentric DID/number the call will not be processed properly and will fail. We do this to prevent callerID spoofing.
You will generally hear the call ring endlessly or simply fail to fast busy signal. We will not accept these calls and they will not show up in your calling reports. In order to avoid this please do not use a Callcentric DID as the calleriD of an incoming call into the Callcentric network.
NOTE: The above also applies to SIP URI calling. |
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| I can't register even after changing my password. What's going on? |
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There may be various reasons for this happening. This FAQ provides more detail on the topic.
Additionally you can check to see whether you enabled the IP White list option. This will be located under the SECURITY tab of your My Callcentric Preferences.
If you are sure you want to use the IP whitelist feature, then make sure you have configured the correct IP addresses. See this FAQ for more information on the IP whitelist feature. |
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| What is the IP whitelist feature and what does it do? |
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The IP whitelist feature is a security oriented tool which allows users to configure IP addresses from which they want to allow SIP registrations. By entering IP addresses the user will be preventing any IP not listed from registering any UA to the account. Registrations will be accepted only from the IP addresses listed.
This feature does not prevent login to your My Callcentric account based on IP.
This feature does prevent SIP registrations to your account based on IP.
This feature is meant for advanced users who use static IP addresses.
You should not use this feature:
- If you use a dynamic IP or dynamic hostname
- If you use a mobile softphone
- If you register your UA from multiple locations
Enabling this feature with no IP addresses in the list will prevent ALL registrations to your account, even your own registration.
You should enable this feature:
- If you use an IP PBX with a static IP
- If you have a static IP
- If you will keep track of the IP addresses you use
If you are not sure of whether to use this feature then please open a trouble ticket on your account so that we may explain whether or not this feature will work for you. |
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| Does Callcentric have peering with other networks? |
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Yes. Any SIP network around the world can call your Callcentric number assuming they allow SIP URI dialing. For details on how you can be reached from outside Callcentric via your SIP URI please see this FAQ.
Callcentric customers can dial to other SIP networks in a few different ways:
1. Directly dial a SIP URI from your IP phone. This normally isn't convenient to do since most phones do not easily support dialing letters ( abc ) and special symbols like period ( . ) and the at ( @ ) symbol.
2. You may use the Phone Book feature available on the MY CALLCENTRIC website to store SIP URI's which can be dialed from Speed Dial numbers like *7501, *7502, etc.
3. You can use the Click 2 Dial feature available on the MY CALLCENTRIC website to enter the SIP URI you would like to call and when you click the DIAL button a call will be placed to your Callcentric phone (or any other number you specified) and once you answer the SIP URI you entered will be called.
4. Use peering numbers. Peering numbers allow you to dial to other networks without having to dial the domain (I.E. @abc.com). Below is a list of peering partners:
**275 - SIP Broker
SIP Broker allows you to call to over 300 VoIP networks using peering codes listed here. To dial via SIP broker you would dial **275 (which is the access code for SIP Broker), then the code of the network (peer), then the number. Here are some examples:
Example 1:
**275*011188888
(Shown separated: **275-*011-188888) This calls to the SIP Broker (**275) Alias Server (*011) to the number 188888 which is the SIP Broker test announcement.
Example 2:
**275*393958
(Shown separated: **275-*393-958) This calls via SIP Broker (**275) to FWD (*393) to the number 958 on FWD which will repeat back your phone number to you.
Example 3:
**275*74712220000000
(Shown separated: **275-*747-12220000000) This calls via SIP Broker (**275) to SIP Phone (*747) to the number 12220000000 on SIP Phone which will put you in the default SIP Phone conference room.
Basically dialing **275 plus the peering number listed on this page plus the number, will get you where you want to go. |
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| Does Callcentric support DTMF? |
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Yes Callcentric does support DTMF.
We support both In-Band DTMF and Out of Band signaling through the industry standard RFC2833 specification.
We recommend sending DTMF through the RFC2833 standard since it provides a more reliable way of transmitting DTMF signals compared to In-Band where compression, especially with highly compressed codecs such as G.729, in combination with poor connection quality can distort DTMF signals and make them unrecognizable.
If you choose to use the In-Band DTMF signaling method and experience problems with sending DTMF tones you can try to solve this by choosing G.711 as your preferred/default codec. This way your signals will be sent but won't be as compressed as when they are sent with G.729, or more highly compressed codecs.
If you are not able to choose RFC2833 standard as your DTMF type in your device, you may also be able to select AVT which is the name used for RFC2833 on some devices. You may also try setting your DTMF signaling type to Auto; this will allow your UA to choose the most suitable signaling type.
Callcentric does not support SIP-Info at this time as a method of transporting DTMF signals.
Here is a short list which explains Callcentric DTMF support for SIP communications:
1 - In-Band: Requires a stable connection and works best with a less compressed codec such as G.711. An unstable connection can still affect DTMF signaling however.
2 - RFC2833/Out of band: Works with any codec, as long as your connection does not suffer from high packet loss/ poor quality
3 - SIP-Info: Not supported at this time
If you still aren't able to send DTMF tones you may open a trouble ticket by logging into your account to contact us. |
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| What is a User Agent (UA)? |
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A user agent is the software or hardware that you use for calling.
Software user agents (or softphones), such as the Counterpath X-Lite softphone, allow you to place and receive calls using your computer, a microphone and headphones. There are also other software user agents which function like IP PBX systems, such as Asterisk, 3CX, pbxnsip...etc. Any SIP/2.0 compatible software is an example of a software user agent.
Hardware user agents are equipment that connect to your broadband internet connection (such as DSL or Cable) and let you place and receive calls using a traditional telephone. Hardware user agents may be IP phones, hardware PBX systems or pretty much any SIP/2.0 compatible hardware..
Please visit our support page for a list of devices (UA) that we have tested with our services, as well as the individual setup guides for those UA. |
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| Why do I get the message "Your phone is not registered" when I log in to my account? |
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If you log in to your account and see the message "Your phone is not registered" on the left side of the page then there could be many reasons to explain why your UA (phone) is not registered.
You can first check to make sure that you have setup your UA properly by visiting our support home page. If your UA is not listed on that page you may follow our Generic / Other setup guide.
You can also make sure that your internet connection is working and that you can ping our servers.
If you are able to ping our servers you can then check to make sure that your ISP is not blocking SIP/VoIP packets.
If you are using X-Lite you can also make sure that the "Register with domain and receive incoming calls" option is checked in your configuration. If you are using a UA which supports DNS SRV then you will want to make sure that you have enabled this feature. You will also want to use public DNS servers (4.2.2.1, 4.2.2.2, 4.2.2.3...etc) in your network configuration.
If all else fails you can easily contact us and we will be happy to assist you. |
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| Can I select which caller ID/phone number is sent on my outbound calls? |
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NOTE: You will NOT be able to use a non Callcentric toll free number as your outgoing caller ID due to FCC regulations. No carrier will accept a toll free caller ID which does not match the actual carrier and owner of said toll free number.
Yes you can select the caller ID you wish to send out per call. Callcentric supports the ability to send one of the caller IDs on your account, instead of only the default one, on each outbound call. This feature primarily applies to business customers and customers running their own IP PBX system. If you do not wish to send out a different caller ID per call then setting your default caller ID in your preferences should be sufficient for you.
Using this feature you may include the caller ID number to send on your outbound call within the SIP (Session Initiation Protocol) INVITE call setup message.
You may pass caller ID numbers for DID's already on your Callcentric account; and you may also pass a caller ID number for number(s) that do not belong to your Callcentric account (such as a cell phone) once you have verified those numbers with us.
This feature is generally useful if you have many extensions behind an IP PBX and wish to pass a unique caller ID (such as the DID) for each extension. It is only available for sending caller ID of numbers in Country Code 1 which includes the USA, Canada, and some other countries primarily in the Caribbean.
NOTE: If you want to pass the caller ID of a number NOT on your Callcentric account you will first need to get your number(s) verified.
To send the caller ID number of a DID on your account or an already verified number within the SIP INVITE message of an outbound call you will need to have your user agent (UA) attach any of the following headers from highest priority to lowest priority, which are supported by most IP PBX's (check your vendor's documentation for support), to your outbound calls:
REMOTE-PARTY-ID
P-ASSERTED-IDENTITY
P-PREFERRED-IDENTITY
When we receive an outbound call from your UA with any of the above headers, and that header includes a caller ID number that is a DID on your account OR has been verified on your account we will pass that as the outbound caller ID for that call; over-riding any settings in your Callcentric account preferences. |
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| Does Callcentric support 7 or 10 digit dialing? |
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Currently Callcentric does not support account level 7 and 10 digit dialing. This is something we may introduce in the future however until it is a stable feature we cannot give an ETA or exact date as to when such a feature may be available.
In the meantime users may configure their devices to support 7 or 10 digit dialing. Listed below are some of the more common devices and software which support 7 or 10 digit dialing:
NOTE: Please be aware that some hardware/software simply DO NOT support 10 or 7 digit dialing while some may only support either 10 or 7 digit dialing.
NOTE: NPA = Your local area code or the area code you want 7 digit dialing for.
NOTE: Please make sure to replace NPA with your desired area code for 7 digit dialing where indicated.
PAP2 and Linksys/Sipura Devices
(*xx.|*xxx|*75xx|0|00|<:1NPA>[2-9]xxxxxx|<:1>[2-9]xxxxxxxxx|1[2-9]xxxxxxxxxS0|011[2-9]x.|[3469]11|**275*x.)
Telco Systems AC-211
(>#|[3469]11|p([2-9]xxxxxx)1NPA|1[2-9]x.T[2-9]x.T|011[2-9]x.T|00[2-9]x.T|**275*x.T|*xx.T|P([2-9]XX[2-9]XXXXXX)1|P([2-9]XXXXXXT)1NPA)
Innomedia SIP MTA-6328
Unfortunately since the Innomedia MTA does not support digit replacement in the dial string it is not possible to configure 7 or 10 digit dialing for this device.
Asterisk/Asterisk Derivatives
Asterisk offers the ability to configure very complex dial plans. Listed below is a dial pattern/rule you can add to your current asterisk Callcentric outbound route to add support for 7 or 10 digit dialing. Please make sure that you are configuring your dial plan properly and that the outbound route to Callcentric is accepting the calls you want to go out through it.
1NPA+NXXXXXX
1+NXXNXXXXXX
pbxnsip
For pbxnsip creating a dial plan is very simple as it can be done easily from the web interface. Note that callcentric is the standard name of the Callcentric trunk and 9 is the standard prefix in our pbxnsip configuration guide. To configure your 7 and 10 digit dial plans simply navigate to your Dial-Plans page and make sure you have the following:
| Trunk | Pattern | Replacement | | callcentric | 9NPAxxxxxxx | 1NPAxxxxxxx | | callcentric | 9xxxxxxxxxx | 1xxxxxxxxxx |
3CX
3CX also offers a very simple interface for configuring dial plans. By default the outbound route for Callcentric uses 9 as the qualifier so we make sure to strip 1 digit from the number dialed for ever pattern we add. Please use the information below to create outbound rules for Callcentric in order to enable 7 and 10 digit dialing:
7 Digit
| Rule Name | Rule for Callcentric 7 Digit | | Calls to numbers starting with (Prefix) | 9 | | Calls from extension(s) | | | Calls to Numbers with a length of | 8 |
| | | Strip Digits | Prepend | | Route | 1 | Callcentric | 1 | 1NPA |
10 Digit
| Rule Name | Rule for Callcentric 10 Digit | | Calls to numbers starting with (Prefix) | 9 | | Calls from extension(s) | | | Calls to Numbers with a length of | 11 |
| | | Strip Digits | Prepend | | Route | 1 | Callcentric | 1 | 1 |
Grandstream (Models: HT286/287, HT386, HT486, HT488, HT501)
Unfortunately Grandstream devices are not able to support 7 and 10 digit dialing properly. These devices allow you to add a prefix for all outbound calls, which would cause *123 to be sent out at 1*123...etc, which would come in handy if you want all of your outbound calls to prefix a 9 for example. However such a scenario would work in a PBX environment where 9 would select the Callcentric trunk. In short these devices do not support 7 or 10 digit dialing at this time. |
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| What ports do I need to use in my UA or firewall? |
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When configuring your user agent (UA) or firewall to work with Callcentric there are certain ports which need to be enabled to avoid quality and/or stability issues. Generally these ports are configured by default; however for users requiring the specific port numbers and protocols please use the information below:
NOTE: You can customize these ports however users who change the ports are generally doing this in advanced environments and will have access to the technical information required.
SIP Ports
Port = 5060
*Port range = 5060 - 5080
Protocol = UDP or UDP/TCP
Direction = Incoming and Outgoing
RTP Ports
The RTP port may vary by UA. If configuring a firewall you will want to configure a range which includes the default RTP port in your UA. A port range of 10 ports may work for most users.
If you will be placing multiple simultaneous calls then a larger range would be required. Users in this scenario will generally know their range requirements and should be able to implement this accordingly.
NOTE: If you are unsure of how many ports to allow then you may simply allow the range between 10000-65535 for RTP data from Callcentric.
Port = Depends on UA
*Port range = Generally a range of 10 ports should suffice
Protocol = UDP or UDP/TCP
Direction = Incoming and Outgoing
*This is for users who may require a port range for their firewall or router
If you still have further questions on this topic please contact us for further assistance. |
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| Are there any special considerations when using Fring? |
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Yes. We've seen an issue in Fring's software (at least version 4.0.0.9) that causes Fring to be unable to register (connect) to our network due to how it responds and sends SIP messages. We are attempting to get this resolved with Fring directly, however as a workaround you can do the following which should resolve this issue:
When configuring Fring to use Callcentric, instead of entering "callcentric.com" within the Proxy field, instead enter "fring.callcentric.com".
Using "fring.callcentric.com" as the Proxy should allow Fring to connect to our network and place and receive calls. |
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| What video codecs does Callcentric support? |
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Video codec support on the Callcentric network is supported entirely as a pass through option, as in we do not interfere or transcode video codecs. Support for video entirely dependent on the UA involved in the calls.
Before reading further please be ware that we do not provide support for video calling to or from PSTN numbers. We will not investigate such scenarios if presented and will inform users of this ability not being supported.
Support for video over SIP is very dependent on two factors:- Codec support between UA
- Bandwidth available for calls
If you are using software with video codec support then the codec used must be available on the called end. If you are still unable to establish video data then we can attempt to assist you, however limited our assistance may be.
For reference below are some of the more popular video codecs used:
H.261
H.263
H.263+
H.264
Some devices may also support other less known video codes such as:
Theora
x264
You will want to disable any unused codecs for compatibility reasons as some codecs may affect call establishment. Ideally choosing a single video codec should be the best option.
Please keep in mind that the support for video calls mentioned above is highly dependent on your internet connection and UA. If you have problems you may contact us for further assistance. |
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| What is a channel? |
|
A channel essentially determines how many calls you can receive, or place, at once. The more channels you have the more simultaneous calls you will be able to have. By default your account comes with the ability to make 3 simultaneous inbound and outbound calls within the Callcentric network, and other SIP networks.
If you purchase paid services then the number of channels you will have is as follows:
Outgoing calling - By default all of our outgoing services come with the ability to place up to 3 calls at once. Note that there is a setting in your account (called "Multiple calls at once") which you will need to set to enable placing multiple calls at once.
We can increase your outgoing calling capacity under the following services:
Pay Per Call
North America 500
North America 1000
Under the above plans we can initially increase your outgoing channels to 3. If you would need greater than 3 outgoing channels you may open a trouble ticket and we would begin the process of increasing your outgoing calling ability.
Incoming calling - Our DIDs each come with a predefined number of channels. On certain DID services you may be able to increase your incoming calling capacity by adding extra channels. You can view a detailed comparison of our DID services here.
NOTE: You will want to have Call waiting enabled in order to receive multiple incoming calls.
So in essence with our paid services you can define how many channels you would like to have for incoming and outgoing calling.
If you have any questions regarding channels or need more help you may contact us for further assistance. |
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| What is a SIP URI? |
|
A SIP URI is an address, similar to an email but NOT an email, used to contact other SIP users on other SIP networks. The standard format for a SIP URI is:
USER@ADDRESS
For example:
[email protected]
A SIP URI should not contain special characters, such as "(), [], {}, *, #, $, !, ^". Using these characters may cause problems as these are special characters and may provide an unintended result.
For a non Callcentric user trying to contact you you directly over SIP may use the format [email protected] or [email protected]. This is described further here. |
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| What is a call treatment? |
|
| A call treatment is an advanced method of number/DID forwarding which allows you to set certain parameters based on time, callerID, called number and other criteria which would determine where your incoming calls get delivered to. Call treatments give you the power to control exactly where your number goes and when you want it to go there. |
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| Where can I send my numbers with call treatments? |
|
You have the option of choosing the exact destination for your incoming calls with call treatments. This can range from a single location to multiple locations using simultaneous ringing or call hunting. For any rule to a single destination you have the following choices:
- Voice mail: Send to voicemail
- Fax: Send to fax
- Calling Card: Send to the calling card feature
- Error message: Send to a Callcentric error message stating that the user is unavailable
- Busy tone: Send to a busy tone
- Error message: Number disconnected: Send to a message stating that the number has been disconnected. This is different from the "user unavailable" error above in that this error states that the number which was dialed itself is disconnected
- Send to my phone: Send to your currently reregistered SIP UA
- Send to my extension: Send to one of the extensions on your account.
- This number: Send to the SIP URI, Callcentric 1777 number or PSTN number (mobile or landline) of your choice
Note: All calls forwarded to a PSTN number (mobile/landline) are billed per minute as described here. Calls to other Callcentric subscribers or other SIP users are not billed. |
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| What is SIP? |
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SIP stands for Session Initiation Protocol. This is the widely used non-proprietary protocol, or language, which our servers use to communicate with your software or hardware. SIP allows the user to initiate or receive calls using video, audio or simple text messaging.
SIP is not specific to Callcentric and many VoIP providers that actually use SIP allow for inter-connection with your Callcentric account to either place or receive direct calls at no extra cost. |
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