Though you can start faster with a softphone, if you are to truly replace your regular landline Bell service with VOIP, you need to set up your ATA.
Your provider is most likely providing personalized settings for your device. Some providers, like CanadaPhoneLine, demand a fee for SIP settings. I have copied and pasted a sanitized version below of my own settings, after a forum post. Show my personal settings (encrypted).
The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:
Dial: **** (That is 4 asterisks)
Once this is done, dial: 110# (110 followed by a square)
The system should now playback the IP Address your device has been assigned.
Using your favourite web browser from a computer on the same network, point the address to the IP address of your adapter.
(example »192.168.1.2) Replace 192.168.1.2 by the IP Address your device is currently using.
You should now see the web interface of your Linksys/Sipura.
click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".
Under the LINE 1 Tab, Find the following fields and fill them with the following information
Display Name: Enter your full name or company name
User ID: REMOVED
Register Expires: 3600
Nat Keep Alive: Yes
Nat Mapping/Traversal: Yes
Step 5 (Optional)
Optionally, To save bandwidth, you can change Line 1 "Preferred Codec" to G729a and make sure "Use Pref Codec Only" is set to no.
Step 6 (Optional)
Optionally, you can configure your adapter with a better dial plan, allowing faster dialling of 10 digits number (Local US/Canada) and also enable 7 digits dialling in one area code of your choice.
At the bottom of Line 1 TAB, you will find a field called Dial Plan
Replace the 416 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:
If you have forgotten your PAP2TN password, you can reset it as such:
- Dial from the telephone attached with device PAP2T ( **** ) four stars
- Now dial 73738# and then press 1 to confirm to factory reset.
- The device will give you buzz sound for reset.
- Start programming from the beginning.
If you can choose from several servers, you would normally choose the one geographically closest. Sometimes however, that server is servicing far too many clients resulting in relative poor performance. In that situation, you might want to run a series of pings, which should provide information on latency.
As expected, the geographically closest server has the lowest latency.
I initially made a $25 deposit and was charged as follows:
- charged to cc 26.25 (incl 1.25 tax)
- Total Setup: $0.50
- Total Monthly Fee: $1.49
- Grand Total: $1.99
They also charge $3.0 for the first month of Enhanced 911 – but unlike other providers, it is optional. There is a $0.0125 charge for CNAME lookups on incoming calls without CID (mostly from USA) but this is also optional. The unlimited incoming calls plan, available only on residential accounts, costs between $5-7, depending on your area code. Other providers may have different plans, but whenever you hear “unlimited”, even though the amounts sound minimal, you may end up paying more than if you were to pay per use in 6s increments.
Charges might vary in time with promotions, etc. If paying by PayPal, consider that they also charge transaction fees of around 3% + $0.30 fixed amount. See a calculator in the links below – though most of the time the fees are absorbed by the provider (payee).
Most of the following tips come courtesy of Mango / TOAO (see the link below). I did not find a need to configure my adapter for most of them, but you might. Most of these settings may only be set using the Advanced Administrator login. To access this, navigate to http://[PAP2T_IP_address]/admin/advanced.
We are all familiar with the good old Voicemail, commercialized by Bell under the name “Call Answer”. Though you can certainly have your own voicemail on your premises, it is preferable to purchase it as a service from your provider, just in case you are somehow completely inaccessible, or if you are getting simultaneous calls while on the phone.
To record a greeting you usually have to login to your voicemail by pressing *97 or *98, then record it in one of the menus. However, if you want a more complex approach, where you give the caller the possibility to call you on other phones as well, or you want to set up your very own calling card (aka DISA), you will need to create an IVR system. For this you need to first design the flow, then create its components, including recording the prompts.
In most cases, the required prompts need to be 22.050 kHz, 16 bit, mono wave (*.wav) files. In Windows, you can use the included Sound Recorder application (see image below for XP). Elsewhere, you can use Audacity which is free and multiplatform. Before exporting your recording, ensure that it conforms to specs.
Another way of obtaining the voice recording wave files is to call your own voicemail and record it as a message, then wait for the system to email the resulting msg#####.wav file to you.
e.g.: SIPdiscount with DLink settings from Deal @ RedFlagDeals
This was posted a long time ago, in 2006, but that service is still available. They offer a number of free destination, but it is unclear how long that could last.
Setting up X-Ten Lite softphone
Though for most households the best bet is undoubtedly an ATA (see above), there is no reason to stop at that. If the provider allows more than one channel – and most do – you can also connect to your account from a softphone and make outgoing calls regardless of whether the main ATA is used or not. For incoming calls the device to have registered last with SIP will ring first.
The following presentation was inspired by fonosip.
On the X-Ten Lite program. Click on the MENU button (on the right of CLEAR and on the left of the green button)
Select System Settings, SIP Proxy, Default. And enter the following info:
Testing the Service
To test the service on fonosip dial 613 for an echo test (A service that test the quality / latency of your connection). Also try 411, 555, 8004, 33091
- If the phone fails to login, please take the time to double check your configuration as above.
- If everything appears to be correct, the problem may be your firewall
- If you are running XP, try disabling the built in firewall.
- If you have an external firewall try opening SIP ports
- If everything else fails and your router / Firewall supports DMZ, try putting the machine with the softphone on the DMZ; try this last, it’s a bad idea
- SIP signalling port (UDP listen) = 5060 RTP/RTCP ports (UDP) = 8000 – 8001
- RTP/RTCP port (UDP listen) = 10000 - 30000 (depending on your sip phone)
Our interesting codes
You might want to call a few of the testing numbers of your provider and see how that works, before calling real people. The following are known to work with my provider:
1-555-555-0911: Test CallerID and e911 Test
044+Country Code+number: International Premium (override account setting – 033 for Value)
*97 Voicemail (*98 for prompt)
4443: Echo Test (will speak back to you – to test how others hear you)
4747: DTMF Test (to ensure that dial tones are recognized)
311: Non-Emergency Police, Municipal and Other Governmental Services (Canadian Servers)
511: Provision of Weather and Traveler Information Services (Canadian Servers)
811: Non-Urgent Health Teletriage / telehealth Services (Canadian Servers)
If you are one of my customers, you may call me or email me your questions at any time. If not, feel free to ask below!