| 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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| 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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| 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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| What is a SIP URI? |
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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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| Where can I send my numbers with call treatments? |
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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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| How do I use simultaneous ringing and call hunting? |
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Simultaneous ringing and call hunting add an extra dimension to your call handling/forwarding options allowing you to deliver your incoming calls to multiple destinations with a single rule. You have the option of delivering your incoming calls to 5 different destinations, with the final destination allowing you to send to voicemail, your calling card, fax or a busy tone. Simultaneous ringing and call hunting are described as follows:
- Simultaneous ringing: Use this option if you wish to have multiple phones ring at once. Your phones will stop ringing after one of the destinations picks up or the specified ring time times out in which case the call will be sent to the 2nd location
- Call hunting: Use this option to have our system search for you by ringing each destination for a specific number of seconds. If the first destination does not pickup then the next destination will be tried until the call reaches the last location
For each destination you may send to either a number, extension or a SIP URI.
You also have the option of setting the number of seconds a call will ring for. Please note that the actual ring tones are not the same as seconds and you will not be able to calculate ring seconds simply based on the number of ring tones.
Finally you can choose to require dialing "1" to accept the call. Please note that you may want to leave this option disabled if you are sending calls to a PBX with an IVR, as the call may timeout since the IVR will not be able to dial "1".
As you can see call hunting and simultaneous ringing offers you many options for the delivery of your call. You no longer need to be tied down to a single location. |
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| Can I select which CallerID to send out for each extension? |
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You can easily select the outgoing callerID for each extension on your account. This will affect calls going to other Callcentric accounts, SIP URIs and the PSTN. Calls between extensions on the same account are handled differently.
To select the outgoing callerID for an extension:
- Login to your My Callcentric account
- Click on the EXTENSIONS link in the navigation menu
- Click Modify for the extension you would like to edit
- Now select the Caller ID to send: value you want for this extension
- Once you are done click Save
NOTE: When you select your 1777 number a default callerID will be sent out for your outgoing calls. |
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