Here at CuteCom, providing excellent customer service and support to you is our #1 priority.
If for any reason there is a problem, we will work to resolve the issue as quickly and professionally as we can.
We make it easy for you to connect with someone if you have a question about your CuteCom service.
Connecting to CuteCom's VoIP network is easy. You will first need a SIP compatible IP phone. We have certified some of the more popular phones here. If you want to get started immediately, you can download a soft phone, like X-Lite while you wait for your hard phone to arrive.
STEP 1
Follow the instructions to configure the phone you are using. Once configured, you should be able to pick up the handset and get a dial tone. This would indicate that the phone is successfully configured and registered with the network.
STEP 2
Your first call should be to the Echo test server. Dial 600. You should hear an explanation of what the echo test is and then a beep. Everything you say will then be repeated back to you as soon as it is received. If you can hear yourself, then you know that your setup is complete.
Now that you are set up, you are ready to make and receive calls!
| Quick Reference Guide | |
|---|---|
| Echo Test - Dial 600 | Test your audio connection to the server. |
| Who Am I - Dial 8400 | Say the phone number you are dialling from. |
| Info - Dial 411 | News, Stocks, Sports, Entertainment, Travel. |
| Voicemail - Dial 8500 | Check your VoiceMail Messages. |
| Voicemail - LAN | Call any local access number and enter PIN, then dial 8500 to retrieve Voicemail. |
| Conference Room - Dial 87 + Room Number | Join a conference call by dialing 87 + 3 digit room number (e.g. 87000) |
| Conference Room - LAN | Call any local access number and enter PIN, then dial 87 + Conference Room Number. |
| Remaining Balance - Dial 8200 | Verify your remaining amount of call credit. |
| Control Panel | Edit your CuteCom account information. |
| Quick Calling Guide | |
|---|---|
| Toll free outgoing PSTN calls | Dial 1 + 800 + seven digit number (also for 888, 877, 866) |
| Continental US and Canada calls | Dial 1 + area code + seven digit number (even when calling within your own area code.) |
| International calls | Dial 011 + country code + city code + phone number |
| Local Access Number with IPX phone | 8600 then CuteCom number then PIN* number |
| Local Access Number with PSTN phone | Call any local access number, then CuteCom number then PIN* number |
* - Pin number can be added by Premium Members in the PIN field in the CuteCom Control Panel.
Our services can be used from anywhere in the world where you can connect to the Internet. Using your Internet connection and an Internet Telephone or Softphone you can make and receive phone calls through our network. We offer both Pay-as-you-go Calling and Unlimited Monthly Plans and all of our plans include Free in-network calling. We offer the ability to receive incoming calls through phone numbers located around the world. This means you can purchase a phone number from another country and receive calls on that number with no long distance fees. This is great for someone with friends and family in a different country as they can make a local call and it will ring to you no matter where in the world you're located. To learn more about our service and how it works, please review this link on our webpage - http://cutecom.net/pages/how-it-works.
Our services can be used FROM anywhere in the world. CuteCom lets you use your Internet connection to make and receive phone calls using our VoIP network. We offer Pay-as-you-go Calling, Unlimited Monthly Plans, and Free in-network Internet Calling along with high quality Internet phones and VoIP devices. To learn more about our service and how it works, please review this link on our webpage - http://cutecom.net/pages/how-it-works.
Our basic rates can be found at the rate calculator listed at this link: http://CuteCom.net/pages/rate-table. Please be sure to check the specific prefixes to determine the costs. Pay-as-You-Go calling which works like a prepaid phone card and lets you call any regular telephone number in the world based on length and destination of the call. You can add additional call credit from your online Control Panel any time and Pay-as-you-go accounts come with all of the CuteCom features. (includes free in-network Internet calling) Monthly Calling plans offer unlimited calling to groups of countries around the world. Choose the USA and Canada or a plan that includes calls to as many as 36 countries. You choose the plan that is right for you based on where you call and what you want to spend. The plans are designed for Individual residential customers or businesses with less than five employees. Includes free in-network Internet calling. CuteCom customers can have CuteCom Direct Incoming Phone Numbers with US area codes in 36 states, as well as London, Toronto, other International locations and USA/Canada toll free 1-800. More International CuteCom Direct Incoming Phone Numbers will become available on an ongoing basis.
Our Long Distance rates are found at http://cutecom.net/pages/rate-table. Please be sure to check the specific prefixes to determine the costs.
All of our accounts have a Call History interface that shows all calls made from each account.
Our Calling Card services were designed as a feature for the existing VOIP customer, so they could use their VOIP minutes when away from their computer and/or VOIP phone. You can create your own PIN number or set PINLESS dialing from the user Control Panel. For more information and to view our calling card access numbers, please visit: http://cutecom.net/pages/local-access-number-list
You can call our local portals to gain access to your voicemail. You can find the portal for your location by visiting the following link: http://cutecom.net/pages/local-access-number-list
Once you have dialed the portal, it will ask you for your 12 digit calling card number. The calling card number consists of your virtual number and a 5 digit pin number. You can set up the 5 digit pin by logging into the control panel and clicking on Calling Card Setup on the left hand side. Once your 12 digit calling card number is entered it will ask you for the number you wish to dial, simply dial 8500 and follow the normal steps for voicemail like you were dialing from your IP/Soft phone.
We offer two services for toll free calls. Unlimited Toll Free and Pay as you go Toll Free. The North America Toll Free Unlimited plan is $54.95 a month, it includes a toll free number that can be dialed from any US/Canada phone, and it includes incoming outgoing calls to/from phone numbers in the US or Canada. Pay As You Go Toll Free Calling allows INCOMING calls on the toll free line billed at $0.04/minute. You will need to maintain a balance of PrePaid Call Credit on the account in order to receive the incoming calls at the incoming rate.
You can add a US/Canada toll free phone number to any existing account through the user control panel. The fee for a toll free number is $4.99 per month and 0.04 per minute.
You can add a real US or Canada phone number to any plan or Pay As You Go account for just $7.99 per month. International Phone Numbers are also available. Check our phone number availability and pricing here: http://cutecom.net/services/did-numbers-0
Problem:
Cannot make calls due to error 306.
Solution:
E911 is an emergency public safety response feature in the USA only. We were required by the FCC - Federal Communications Commission of the US to warn and receive client acknowledgment that the E911 over VoIP is limited.
You have to go to your User Control Panel and login with your Virtual Number and case sensitive password. Once you have made that acknowledgment, you will be able to proceed to use the service.
Problem:
Can I send a fax from one user phone number to another (IP to IP)?
Solution:
This FAX service is not guaranteed.
IP to IP faxing, both the sender and receiver need to be CuteCom members and have their fax machines attached to IP analog telephone adapters (ATAs).
One other important factor when faxing over VoIP is the quality and speed of your Internet connection. Since fax data cannot be compressed, the G711u/a CODEC must be used. This codec requires a minimum of 64Kb/s in both directions to be reliable, but we recommend more than 90Kb/s.
Problem:
Can I send a fax from my account to a traditional fax machine(IP to PSTN)?
Solution:
IP to PSTN faxing is not supported.
We support G729a, G711u/a, G723, G726, GSM and ILBC codecs.
To dial any PSTN phone number that is part of the NANP (North American Numbering Plan), you must dial:
1 + Area Code + Phone number
The NANP covers the United States, Canada, Bermuda and most Caribbean countries. Despite the "North American" name, Mexico and the Central American countries are not part of the system.
Note: PSTN numbers must be dialed as if you were living in North America. If you reside in Australia and need to call Canada, United States, Bermuda or most Caribbean countries, you will need to dial: 1 + Area Code + Phone number.
To dial any PSTN phone number that is outside of the NANP (North American Numbering Plan), you must dial:
011 + Country Code + City Code + Phone number
The Residential unlimited plan is based on the calling habits of the average individual and is only for use in a home or other residential setting. CuteCom retains the right to require account upgrade or suspend service if the calling habits are outside these guidelines.
The Business unlimited plan is based on the calling habits of an average small office user sold on a per seat basis. Business plans are designed for businesses who are not in the business of making and taking phone calls. Businesses that require multiple users making simultaneous calls or high volume users in the business of making or taking phone calls should use our Bulk Plans.
If the country you call most is not offered in an unlimited plan, you can still save money. The type of account that you would need to order is the Pay-As-You-Go account. This plan works like a prepaid phone card and lets you call any regular telephone number in the world based on length and destination of the call. You can add additional call credit from your online Control Panel any time and Pay-as-you-go accounts come with all of the CuteCom features. (includes free in-network Internet calling) Our basic rates can be found at the rate calculator listed at this link: http://CuteCom.net/pages/rate-table. Please be sure to check the specific prefixes to determine the costs.
Bulk Plans offer options for call shops, telemarketing firms, Internet cafes and other high volume users that want to have more than one user using the service at one time. Bulk Plans allow users to have many simultaneous calls and share the balance of minutes.
Problem:
What local SIP port do I use when more than 1 SIP device is in use?
Solution:
A well designed router will be able to cope with use of the same local SIP port 5060 but most residential routers cannot. Depending on the device, this port is usually called SIP port or local SIP port and needs to be adjusted to reflect a different port number per device line. You can start at 5060 and work your way up using even number, skipping odd. For example you have 1 PAP2, 1 SPA and 1 Xten. Allow the PAP2 to use 5060-5062, change the ports on the SPA to 5064-5066 and assign the Xten SIP port 5068.
You can hear your Call Credit Limit Balance by dialing 8200 from any Virtual Number.
Registration is the event where your IP Phone (UA) contacts our system and tells it where you can be reached for incoming calls. The registration interval is the time (measured in minutes or seconds) between when your UA re-registers. Registration is required before you can make PSTN calls.
Our sip proxy forces a 60 minute registration regardless of the interval specified in your VoIP adapter.
The unlimited plans are designed for the calling habits of the typical residence. In some cases, we find certain users have requirements over and above what the industry accepts as typical residential usage. You fit into that category of users. When pricing our plans, we have to consider the average consumer and cannot allow usage that would be typical for a business, call center, telemarketing firm or other high volume user. As per the Terms and Conditions found in our General Service Agreement, we ask that you upgrade to a bulk plan or pay as you go account immediately. In order to upgrade, please place an order for the new service. Once this is complete, let us know the new account and the existing account that needs to be deleted.
Make sure to have the SPA9000 IP address and the corresponding port, the default port is 6060.
For example 192.168.0.1:6060 is put in the Proxy location of the SPA941.
You should also check if you are connecting the SPA941 to the correct interface LAN or WAN of the SPA9000.
Problem:
Will VoIP work with a Satellite Internet connection?
Solution:
In general, we do not recommend the use of our system over a Satellite (SAT) Internet connection. However, some customers do. There are a few technical issues related to SAT links that can cause problems:
- Round Trip Time (RTT) Delay. Most SAT's used for Internet connections are geosynchronous. This means that their orbit period is the same as the length of a day, so they are stationary in relationship to a given point on the ground. To achieve such an orbit, the SAT must be 22,235 miles (35,784 km) above the earth's surface. Radio waves travel at the speed of light, but even the speed of light takes a noticeable amount of time to travel that distance 4 times (up and back twice for a complete round trip from your UA to our servers.) This delay, when added to other factors on the ground, can cause delays anywhere from 500 ms to 1000 ms (1/2 second to 1 second.) It can be very difficult to carry on a conversation when there is that much delay.
- Network Jitter. Probably more important than RTT Delay itself is the consistency (or lack of) of the RTT Delay. Some packets take longer than others to reach their destination which can cause voice data to arrive out of order and possibly be dropped. We have found that many SAT Internet connections have this failing. Many UA's have adaptive Jitter Buffers which can absorb some small inconsistencies in RTT delay, but given large inconsistencies, the result is often an almost unusable connection.
- Lack of uplink bandwidth. Typically SAT Internet connections are asymmetrical with their download bandwidth being much higher than their upload bandwidth. If your uplink bandwidth is lower than the bandwidth required by the vocal codec you are using then you will not be able to use the connection for voice calls reliably.
Problem:
Can I send a fax from one user phone number to another (IP to IP)?
Solution:
This FAX service is not guaranteed.
IP to IP faxing, both the sender and receiver need to be CuteCom members and have their fax machines attached to IP analog telephone adapters (ATAs).
One other important factor when faxing over VoIP is the quality and speed of your Internet connection. Since fax data cannot be compressed, the G711u/a CODEC must be used. This codec requires a minimum of 64Kb/s in both directions to be reliable, but we recommend more than 90Kb/s.
Problem:
Can I send a fax from my account to a traditional fax machine(IP to PSTN)?
Solution:
IP to PSTN faxing is not supported.
Problem:
What digits do I need to dial to make an International phone call?
Solution:
To dial any PSTN phone number that is outside of the NANP (North American Numbering Plan), you must dial:
011 + Country Code + City Code + Phone number
Problem:
How fast does my connection to the Internet need to be in order to use the service?
Solution:
Your connection speed (bandwidth) must be sufficiently high for the type of vocal codec you are using. Vocal codecs (or just "codec") are the standard means by which your IP Phone (UA) encodes your voice and transmits it across the Internet. The codec that is picked for a particular phone call is primarily determined by your UA's configured preferences. Consult your UA's manual to determine how to change codec preferences.
To use the best quality / highest bandwidth codecs (G711a/u) we recommend a connection speed of no less than 90Kb/s in each direction (up and down).
PSTN is an acronym for Public Switched Telephone Network. It is the traditional telephone system that the world has been using for decades to make phone calls. In order to access the PSTN from your IP Phone (UA), you need to use a "bridge" between VoIP and PSTN.
If you are a Premium Member, this is accomplished seamlessly by dialing a 1 + areacode + phone number for a North American call or 011 + country code + city code + phone number for an International call.
Problem:
Sometimes when I try to place a PSTN call, I get a recording stating:
"An error has occurred. Please contact Technical Support with error code 104."
Solution:
The problem is that the system has detected more than one IP Phone (UA) registered with a phone number that is associated with an unlimited calling plan. Calling plans are restricted for use by only a single UA to prevent abuse. The simple solution is to make sure that you only have one UA programmed with the virtual phone number that is associated with your calling plan.
Occasionally, due to power outages or UA reboots more than one registration can appear for the same UA, triggering an error 104. The old registration will eventually expire (you can check for when on your control panel.)
Problem:
Sometimes when I try to place a PSTN call, I get a recording stating:
"An error has occurred. Please contact Technical Support with error code 205."
Solution:
The system has detected a PSTN call from an IP Phone (UA) that is not registered correctly. To prevent abuse, all UA's must register with our proxy server before they can place a PSTN call. Common reasons for your UA not to be registered are:
PROBLEM: Incorrectly set USERID (phone number) or PASSWORD in the UA
SOLUTION: Make sure your USERID and PASSWORD are set correctly. Re-enter them if necessary.
PROBLEM: "Registration" is set to "Never" or "No" in the UA's configuration
SOLUTION: Modify your UA's configuration to force it to register before placing a call.
PROBLEM: Call placed too soon after registration
SOLUTION: Wait at least 5 seconds after successful registration before making a PSTN call.
Registration is the event where your IP Phone (UA) contacts our system and tells it where you can be reached for incoming calls. The registration interval is the time (measured in minutes or seconds) between when your UA re-registers. Registration is required before you can make PSTN calls (see the Knowledge Base article on Error 205.)
Our sip proxy forces a 60 minute registration regardless of the interval specified in your VoIP adapter.
Problem:
Sometimes the voice of the person I am speaking to sounds choppy or broken up. Why? What can I do?
Solution:
This can be caused by a variety of problems. The most common is poor Internet connection quality or lack of bandwidth.
Very often what is sold as a "High Speed" or "Broadband" connection is not up to the task of transmitting voice packets in a timely or reliable manner. Problems such as high latency (slow delivery of packets) or packet loss (actual dropping of voice data) can be seen from time to time on some provider's networks. These problems will be heard by you on your IP phone call long before they are noticed for other data activities like checking email or web browsing.
The Linksys Phone Adapter enables use of our high-quality feature-rich telephone service through your cable or DSL Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone jacks to connect your existing phones. Each phone jack operates independently, with separate phone service and phone numbers -- like having two phone lines. With CuteCom, you'll get clear telephone reception, even while using the Internet at the same time for normal data operations.
STEP 1
You must first determine what IP address it received. To do this, you need to pick up the phone attached to the Line 1 jack and
dial: **** (four asterisks)
then dial: 110 #
and you will be told the IP address of your device (e.g. 192.168.0.100).
STEP 2
Go to any browser equipped computer on your network and enter the address:
http:// <IP ADDRESS> /
(where <IP ADDRESS> is replaced by the address that was given to you in STEP 1.
STEP 3
Click on the "Admin Login" button near the top right side of the screen, then click on the "Line 1" tab.
STEP 4
You need to modify only a few parameters from the factory default. They are listed here:
| Proxy | sip.cutecom.net |
| Display Name | Enter your Full name. This will show up as part of your caller-id |
| User ID | Enter your CuteCom Phone Number |
| Password | Enter your password for CuteCom Phone Number |
| Register Expires | 3600 |
STEP 5
To save bandwidth, you can change Line 1 "Preferred Codec" to G729a. Also change the "Use Pref Codec Only" to No. You can only do this for one line. So, if Line 1 is on G.729a, Line 2 has to be some other codecs.
STEP 6
Click on the "Save Settings " button at the bottom of the form.
STEP 7
Make calls!
The Linksys Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analog phones or use one of the ports for a fax machine. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and a reliable fax connection, even while using the Internet at the same time.
STEP 1
You must first determine what IP address it received. To do this, you need to pick up the phone attached to the Line 1 jack and
dial: **** (four asterisks)
then dial: 110 #
and you will be told the IP address of your device (e.g. 192.168.0.100).
STEP 2
Go to any browser equipped computer on your network and enter the address:
http:// <IP ADDRESS> /
(where <IP ADDRESS> is replaced by the address that was given to you in STEP 1.
STEP 3
Click on the "Admin Login" button near the top right side of the screen, then click on the "Line 1" tab.
STEP 4
You need to modify only a few parameters from the factory default. They are listed here:
| Proxy | sip.cutecom.net |
| Display Name | Enter your Full name. This will be shown as part of your caller-id |
| User ID | Enter your CuteCom Phone Number |
| Password | Enter your password for CuteCom Phone Number |
| Register Expires | 3600 |
STEP 5
To save bandwidth, you can change Line 1 "Preferred Codec" to G729a. Also change the "Use Pref Codec Only" to No. You can only do this for one line. So, if Line 1 is on G.729a, Line 2 has to be some other codecs.
STEP 6
Click on the "Save Settings " button at the bottom of the form.
STEP 7
Make calls!
The Linksys SPA8000-G1 8-Port IP Telephony Device is a full featured Analog Terminal Adapter (ATA) for small business enterprises providing enhanced communication services via a broadband connection to the Internet.
The SPA8000 features eight RJ-11 FXS ports to connect analog telephones to IP-based data networks and includes a single multi-port RJ-21 50-pin connector offering an alternative connection choice when deploying the telephony gateway in varied environments. The device also has one 10/100Base-T RJ-45 Ethernet interface to connect to either a router or multi-layer switch.
STEP 1
Connect one end of the Ethernet cable (included) to the ETHERNET port of the SPA8000. Connect the other end to your Cable/DSL modem.
Connect one end of a different Ethernet cable to the AUX port of the SPA8000. Connect the other end to the Ethernet port (network card) of your PC.
Power on the cable/DSL modem and then power on the SPA8000.
STEP 2
Make sure your PC is setup to obtain an IP address automatically. This will allow your PC to obtain an IP address from your SPA 8000.
To verify that your PC obtained the IP address correctly, run the IPconfig command at the command prompt.
If for any reason your PC has not obtained the IP address from the SPA8000 even though it is setup to obtain an IP address automatically, run the following commands from the command prompt:
To release any IP address from your network card:
Ipconfig/release
To renew the IP address:
Ipconfig/renew
STEP 3
Open your internet browser and type the following: http://192.168.0.1 and press enter.
Enter “admin” for the user and “admin” for the password.
STEP 4
A setup page will appear. Click on “WAN Setup”.
Choose the Internet connection type depending on how your ISP assigns you your Internet Protocol (IP) number. When you have finished your settings, click on “Submit Changes”. When your Router Setup screen comes back up, you should be able to browse the Internet from your PC. If not, contact your ISP.
STEP 5
Choose from the main menu “VOICE” and then “L1”.
Enter your CuteCom Virtual Number into the User Id field and the CuteCom sip password into the Password field.
Repeat step 5 for L2 – L8.
STEP 6
For the best voice quality, choose codec G729 and then select AVT for the DTMF method.
STEP 7
Submit all changes.
STEP 8
Make calls!
The Linksys SPA941 IP telephone can be configured as a two (2) line or, via a simple software upgrade, a four (4) line full featured business phone with pixel based graphical display, speakerphone and headset port. Stylish and functional in design, the SPA941 can be used in residential, SOHO, enterprise and small to medium business service offerings including IP PBX, hosted IP telephony and IP Centrex. Out of the box, it is configured for DHCP. This means that it gets its IP address from your DHCP server automatically.
NOTE: We have seen instances where installing the SPA941 behind a firewall which blocks ICMP packets causes problems with registration. Try turning off any ICMP blocking on your firewall.
STEP 1:
You must first determine what IP address it received. To do this, on the phone dial pad, press the "envelope" button and then press the button under the LCD screen word "save".
STEP 2:
Go to any browser equipped computer on your network and enter the address: http:// <IP ADDRESS> /
(where <IP ADDRESS> is replaced by the address that was given to you in STEP 1).
STEP 3:
Click on the "Admin Login" button near the top right side of the screen, then click on the "Ext 1" tab.
STEP 4:
You need to modify only a few parameters from the factory default. They are listed here:
| SIP Port | 5066 |
| Proxy | sip.cutecom.net |
| Register Expires | 3600 |
| User ID | Enter your CuteCom Phone Number |
| Password | Enter your password for CuteCom Phone Number |
| Preferred Codec | G729a |
STEP 5:
If the second extension is needed, click on Ext2 and repeat Step 4. Please make sure to increment the SIP port by one. For example, Ext1 SIP port: 5066; Ext2 SIP port: 5067.
STEP 6:
Make calls!
The Linksys SPA942 IP telephone can be configured as a two (2) line or, via a simple software upgrade, a four (4) line full featured business phone with pixel based graphical display, speakerphone and headset port. Stylish and functional in design, the SPA942 VoIP telephone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA942 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP Phone that is unparalleled in features, value, and support.
NOTE: We have seen instances where installing the SPA942 behind a firewall which blocks ICMP packets causes problems with registration. Try turning off any ICMP blocking on your firewall.
STEP 1:
You must first determine what IP address it received. To do this, on the phone dial pad, press the "envelope" button and then press the button under the LCD screen word "save".
STEP 2:
Go to any browser equipped computer on your network and enter the address: http:// <IP ADDRESS> /
(where <IP ADDRESS> is replaced by the address that was given to you in STEP 1).
STEP 3:
Click on the "Admin Login" button near the top right side of the screen, then click on the "Ext 1" tab.
STEP 4:
You need to modify only a few parameters from the factory default. They are listed here:
| SIP Port | 5066 |
| Proxy | sip.cutecom.net |
| Register Expires | 3600 |
| User ID | Enter your CuteCom Phone Number |
| Password | Enter your password for CuteCom Phone Number |
| Preferred Codec | G729a |
STEP 5:
If the second extension is needed, click on Ext2 and repeat Step 4. Please make sure to increment the SIP port by one. For example, Ext1 SIP port: 5066; Ext2 SIP port: 5067.
STEP 6:
Make calls!
Bria is the newest softphone application from CounterPath, allowing users to enjoy multimedia communications in a dynamic new way. Featuring an intuitive new interface, Bria is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages. You can purchase Bria and find full documentation at Bria's website.
NOTE: Do not try to use the # key to send a call as it will be interpreted as part of the phone number. Use the green phone symbol or the enter key instead.
STEP 1:
Click file from the menu along the top of the program and choose Account Settings
STEP 2:
Highlight the Account 1 line and click Edit.
Click Apply.
STEP 3:
Please enter your information in "Accounts Page" as provided below.
| Display Name | Enter your Full name. This will show up as part of your caller-id |
| User Name | Enter your CuteCom Phone Number |
| Password | Enter the password of your CuteCom Phone Number |
| Authorization User Name | Enter your CuteCom Phone Number |
| Domain | sip.cutecom.net |
| Register & Receive Incoming Calls | Checked |
| Send Outbound via | Proxy (Address: sip.cutecom.net) |
Click Apply.
STEP 4:
Click on the Topology tab. Select "Use Local IP Address". Select "Use specified server" and leave the corresponding box blank. Uncheck "Enable ICE".
Click OK and then Close.
STEP 5:
Make calls!
This is one of the better SIP video soft phones that we have tested. This is the quickest way to get up and running with CuteCom's SIP video service. You can purchase and download it and find full documentation at Counterpath's website. Note: by referring you to this 3rd party site, we are neither encouraging you nor endorsing this product.
NOTE: Do not try to use the # key to send a call as it will be interpreted as part of the phone number. Use the green phone symbol or the enter key instead.
STEP 1:
When you have downloaded the eyeBeam Video Phone, click on the arrow on the left side of the software phone. Right click on the wrench symbol and the click on “Settings”. The Settings window will open.
STEP 2:
Click on the Server settings. Put a check mark on the square box on the left of “Enable this SIP account”.
Complete the following fields and then click on "Apply".
| Display Name | Enter your Full Name, this will show up as part of your Caller-id |
| User Name | Enter your CuteCom Phone Number |
| Password | Enter your password for CuteCom Phone Number |
| Authorization User Name | Enter your CuteCom Phone Number |
| Domain | sip.cutecom.net |
STEP 3:
Click on Firewall/NAT settings. Make sure Send Internal IP is set to "Default" and Enable ICE is unchecked. Click on "OK".
STEP 4:
Make sure that your webcam has been installed. You should be able to see your video on the lower side of the left ear of the eyeBeam phone. If you do not see the video, it means that your webphone has not been installed properly.
STEP 5:
At this time, you should be able to dial another video phone that is on line. Simply just dial the number that you want to call. When the callee answers, click on the green “Start” button and you should be able to see the callee's video on top of your own.
STEP 6:
Make calls!
NOTE:
Free soft phones typically only support CODEC G.711. Certain network routes do not support CODEC G.711. This may make it necessary to upgrade the soft phone, or purchase an IP device.
This is one of the better soft phones that we have tested. The lite version is fairly full featured and the price is right, FREE!
This is the quickest way to get up and running with CuteCom's service until your hard phone arrives or if you're choosing to only use your computer for calls.
NOTE: Most of the routes around the world only support a single codec such as G.729 or G.723. Free soft phones typically only support codec G.711. Because of this reason, it may make it necessary to upgrade the soft phone to a paid version that supports codec G.729 and/or G.723, or purchase an IP device.
NOTE: Do not try to use the # key to send a call as it will be interpreted as part of the phone number. Use the green phone symbol or the enter key instead.
STEP 1:
Right click on any part of the softphone and select “SIP Account Settings” Click on "Add".
STEP 2:
Please enter your information in the "Accounts Tab" page as provided below.
| Display Name | Enter your own display name |
| User Name | Enter your CuteCom Phone Number |
| Password | Enter password for your CuteCom Phone Number |
| Domain | sip.cutecom.net |
| Register with Domain and Receive Incoming Calls | Checked |
| Send Outbound via | Proxy (Address: sip.cutecom.net) |
Click Apply.
STEP 3:
Click on the Topology tab. Select "Use Local IP Address". Select "Use specified server" and leave the corresponding box blank. Uncheck "Enable ICE".
Click OK and then Close.
STEP 4:
Make calls!
This configuration guide assumes that you have already followed the main setup guide and have the softphone installed and working with your voip line.
STEP 1:
Click the arrow found on the right hand side of the phone to display the advanced options menu.
STEP 2:
Now click on the CONTACTS tab and you will now see the basic list of Friends, Home, and Work. Right click on one of these categories and choose the Add Contact option.
STEP 3:
Fill in the personal information for your contact.
Under the CONTACT METHODS area, type the person's 7 digit virtual number and proxy they are using into the softphone category. EXAMPLE: 1234567@sip.cutecom.net
Place a check mark into the "Show this Contact's Availability" box and click OK.
Your contacts name will now show up in the list you chose in gray.
STEP 4:
From the menu options (or by right clicking on the display) choose SIP Account Settings.
STEP 5:
Click Properties of your account.
Click the Presence tab and choose Presence Agent. Click OK, then close.
Your contact, if online, will now show up in green.
Problem:
I cannot access VM via the PAP2/Sipura
Solution:
If you find that you cannot access the VM system even with the correct VM password, the SPA/PAP2 DTMF settings are not correct.
Follow the instructions below:
Access the device's configuration settings. If you do not know how to access the device's settings, please use our set up instructions here.
Problem:
If you can make calls but not receive them and the softphone is correctly registering with the sip proxy, your installation might have not been entirely completed.
Solution:
Make sure that every time you install any softphone in Windows 200 or Windows XP you are logged on as the computer Administrator, which will give you full control NTFS permission to the folders needed to access during the installation.
Do not install the softphone using the "Run As" option; this option is not meant for installations but to run a program.
If you use this (Run As) option you might not get an error message; but due to insufficient NTFS permissions for some folders, some components won't be installed, causing this application to fail.
Check the account CP to verify the device connected to the number is registering with the system. Next to the number you will see a line containing: Registration expires @ ... UA Type: type of SIP device.
-If the registration line is missing, the device is not connecting to the system and therefore it will not receive calls. Thus the system will immediately detect the lack of registration and reply with the "person is not available to take your call at this time". In such case, the device is either offline because it is turned it off or there is a problem with it. If the device is not off, the connection to the network should be checked as well as the device configuration settings should be reviewed. A network may be in order.
-If the registration line is present, the following needs to be verified:
a. check call forwarding-CF for that number. If CF is set to "Send immediately to my voice mail" the system will immediately send the call to VM. The send to VM call forwarding option neds to be removed in order for the device to receive calls normally.
b. the device is having connection issues where the registration interval is so long that the device is offline and the control panel registration is still present. The device registration interval needs to be changed to 300 seconds or 5 minutes.
c. the device has problems with the router, NAT issues:
If the router lost connection to the Internet, most routers will flash for an instant and have to reconnect to the attached devices. As a connection is re-established the router will block outside connections to an internal device unless the internal device makes contact first.
Please consider forwarding the device's ports on the router so it will not cut off your SIP device from outside connections.
If you reboot the SIP device, it establishes contact with the SIP server and then the router opens up that link and allows two way communication between the devices. If the router looses contact with the SIP device, then when the SIP server tries to make contact with the device, the router blocks the connection, hence busy signal.
As with all NAT sessions unless you have port forwarding setup the sessions need to be established from the inside. The problem is if the router resets or has an issue then any existing sessions normally get forgotten about and thus dropped. The SIP protocol deals with this by having a reregistration period.
At InPhonex this reregistration period is 5 minutes or 300 seconds. This normally means that if the SIP device is working correctly and the router does lose its information then the SIP device should fix that issue within 5 minutes.
Putting the SIP in the DMZ on your router should fix this problem as well.
Note, the system will allow IP to IP calling without registration/authentication. The system will not allow incoming or outgoing PSTN calls without proper device connection to the system.
If you have a firewall in your modem or router, you must give the Phone Adapter rights to access certain Internet ports, including:
5060 UDP
10,000-20,000 UDP
NOTE: Some devices may need to use 5060 TCP, and also depending of the configuration/device used, different ports may be accessed. It's recommended that you put the device IP on the router DMZ. (This is only recommended for UA devices, not computers using softphones)
Problem:
When a user disconnects a call and quickly attempts to make another call, the first call is not disconnected and billing continues on both calls. In actuality, the first call is put on hold where billing continues while the second call is running simultaneously.
Solution:
Most SIP devices support simultaneous calls via the same line to enable the user to utilize "hold" type functions. Such features like three way calling and call waiting require a call to be placed "on hold" while the second call is initiated. You can disable such functions by accessing the device configuration settings and turning off the service that offers the ability to "flash" and get the stuttered dial tone that allows for a "three way call" or "call waiting". Once you disable these settings, this problem should not re occur. If you are not sure how to do this, please review the device's manual for exact instructions.
The SIP proxy server no longer accepts or returns ICMP packets. This is normal and does not indicate that the service is down.
Problem:
My number is not registered
Solution:
1. Check the following items.
Follow our online configuration guide and Make sure you have the correct VN and password
2. Disable any personal firewall.
Personal firewalls like Norton Personal Firewall could be blocking sip ports from 5060 to 5069. Blocked ports can also cause one way audio.
3. Disable Windows XP firewall
Windows XP has a built-in firewall that could also be blocking these ports.
In order to verify if your Firewall is enabled follow these steps:
If you do not want to Disable your Windows XP Completely:
There is no Internet connection available for the SIP device to use.
Problem:
Possible causes and solutions for Login timed out error.
Solution:
"Login timed out" error is released if the CuteCom number is not correctly typed, there are no dashes. The 7 digit number must be entered as XXXXXXX.
This is most often caused by an incorrect proxy address or password for your account. The proxy should be "sip.CuteCom.com". This error may also be caused by poor Internet connection, unstable network, lengthy registration interval or blocked firewall.
To verify this is a blocked firwall issue, please follow the steps below:
1. Turn off the softphone and access www.testyourvoip.com
2. Run the test to Boston.
3. Please make sure the applet does not use a port other than 5060 to run the test. If the applet needs to use a different port, it will release a message.
4. If the applet allows you to complete the test, please let us know the scores.
If you can not run the test due to firewall settings interfering, you will need to set up the firewall to allow the softphone full access to the Internet or turn off the firewall so the softphone can register with the SIP proxy.
Message waiting indication- MWI service
The device will ring when new messages are waiting in the VM box of that number. The ring will not go away until the user accesses VM at 8500, retrieves and deletes the VM messages.
The device controls the MWI service.
If you do no wish to subscribe to this service, please modify the following settings:
For Sipura and LinkSys devices: access admin/advanced/all lines/Supplementary service subscription:
-MWI service: no
-VMWI service: no
For Grandstream devices: access advanced configuration settings:
SUBCRIBE for MWI: No, do not subscribe for MWI.
To adjust the existing MWI settings for Sipura/LinkSys devices: access admin/advanced/all users/ring settings:
- VMWI Ring Splash Lenght: Duration of ring splash when new messages arrive before the VMWI signal is applied (0 – 10.0s) Default is set to .5
- VMWI Ring Policy The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the SPA indicating the status of the subscriber’s mail box.
3 settings are available:
New VM Available – ring as long as there is 1 or more unread voice mail
New VM Becomes Available – ring when the number of unread voice mail changes from 0 to non-zero
New VM Arrives – ring when the number of unread voice mail increases
- New VM Available Ring On No New VM- If enabled, the SPA will play a ring splash when the VM server sends SIP NOTIFY message to the SPA indicating that there are no more unread voice mails.