VoIP.ms Wiki - User contributions [en]

Cisco Linksys PAP2

2011-07-13T20:59:45Z

Martinm:

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]
<div style="width:100%;overflow:hidden;clear:both"></div>
==Configuration Details==

'''Step 1'''

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)'''<br>
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned.<br>
(Example: 192.168.1.2)

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.'''<br>
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

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".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Proxy:''' atlanta.voip.ms (Atlanta is just a sample, you can put there any of the multiple VoIP.ms servers)<br>
'''Display Name:''' John Smith (your name or company name)<br>
'''User ID:''' 100000 (your VoIP.ms username)<br>
'''Password:''' ******** (the account password)<br>
'''Register Expires:''' 180<br>
'''Proxy Fallback Intvl:''' 180<br>
'''Nat Keep Alive:''' Yes<br>
'''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 dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 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:

(<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|911S0|822|0|00|[2-9]xxxxxx|4XXX|**275.|xxxxxxxxxxxx.)

'''Step 7'''

Click on the "Save Settings " button at the bottom of the form.

'''Step 8'''

Switch to the '''SIP''' tab and scroll down to '''RTP Parameters''' and set the follow setting:

'''RTP Packet Size''': 0.020

=='''Configuration Screens'''==

[[File:PAP2T001.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2 Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2 Configuration Screen|frame|left]]

[[File:Pap2sip.jpg|PAP2 Configuration Screen|frame|left]]

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'''Customer Submitted Information:''' <br>
For North America:<br>
Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with VOIP.ms.<br>
Read the article called: [http://www.toao.net/25-linksys-ata-configuration Configure your Linksys VoIP ATA the right way!]

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2. Changing the PAP2's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the PAP2's web interface.
*2- Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.

[[File:Regional.jpg|800px]]

*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

Phone book

2011-07-10T22:46:27Z

Martinm:

The Phone Book feature allows you to configure Speed-Dial entries and Caller ID name (CNAM) overrides. For example, let´s say you have a Customer, Provider or Relative that you call frequently you can create a phone book entry in order to make call using an speed-dial entry of less than 7 digits, or you can have a Caller-ID name (CNAM) override to identify the calls of an important customer if his number doesn't have a proper Caller-ID name (CNAM) linked to it.

This guide will help you to configure and learn how to use properly the Phone Book.

__TOC__

== Phone Book Purposes ==

Before creating the phone book entry, we are going to explain the main purposes of this feature:

=== Speed-Dial ===

You can create a Phone Book entry to work as an Speed-Dial, allowing you to place a call by pressing a reduced number of keys. This function is particularly useful if you dial certain numbers on a regular basis. You can program an speed dial to local, long distance or international numbers, also you can have a SIP URI (like *7501 to dial john@other-sip-provider.com).

The prefix to dial your speed dial entries is *75. Example: If you want to dial entry 01, you need to dial *7501 from your phone. It's not currently possible to use a different prefix.

=== CallerID-name (CNAM) Override ===

When you receive an incoming call to one of your numbers that matches a phone number in the phonebook, the Caller-ID name of the incoming call can be set in the entry on the phone book. For example, let´s say that you have an entry with the number 5554443322 associated with the name "John Smith", whenever you receive a phone call for that number you will see the Caller-ID name as "John Smith". The phone book feature overrides the caller ID name look up feature (if you have enabled it for your DID number)

=== CallerID Number Override ===

When you call one number using the Speed-Dial if you have setup a "CallerID Number Override" the default CallerID number will be changed for the one you have configure in your phone book entry.

== Setup a Phone Book ==

First go to your Customer Portal and follow the menu option "DID Numbers" >> "Phone Book"

[[File:Phone book.jpg|800px]]

From this screen you can create, edit and delete Phone Book entries.

=== Create a Phone Book Entry with a Phone Number ===

To create an entry, first click on the button "Add Phone Number", you will be prompt to the next screen:

[[File:Pb entry.jpg|800px]]

'''Speed Dial''': Here you only need to select which Speed-Dial want to assign to this number. ''Optional Field.''

'''Name''': You can enter here the name that you want to use as CNAM Override. ''Mandatory field.''

'''Phone Number''': Enter here the phone number. ''Mandatory field.''

You can also use an International Number, you only need to make sure to use the prefix 00 or 011 to make the call to the number. Make sure that you have enable the International Calls in your account.

Note: The CallerID Number is not 100% guaranteed to be passed properly for International Routes at the moment.

'''CallerID Number Override''': Enter here the number you want to pass. This overrides the default callerID Number. ''Optional field.''

'''Note''': Here you can set a note to identify the entry. ''Optional field.''

=== Create a Phone Book Entry with a SIP URI ===

Creating a Phone Book Entry for a SIP URI is basically the same as creating the entry for a Phone Number. The difference is that you can select or create the SIP URI entry in this point. The information is selected for the [[SIP URI]] feature within your Customer Portal.

[[File:Pb entry sipuri.jpg|800px]]

'''SIP URI''': Here you can either select '''Use Existing''' or '''Create New''' to assign the SIP URI to your Phone Book entry.

=== Import or Export the Phone Book ===

[[File:Imp exp pb.jpg]]

Additionally you can import or export your Phonebook, the format is CSV (Comma-separated values). Here's an example of the information that will be look like, when you export your Phonebook.

*7501,"John Smith",5554443322
*7502,"John Doe",johndoe@other-provider.com

'''Note''': If you mark the box '''Overwrite Existing Phonebook Entries''' while Importing your Phone Book this overwrite all the entries with the same information.

PBX Security

2011-05-03T14:03:17Z

Martinm:

Based on the broad view we have of thousands of customers, leads us to believe that most of the hacking cases for the purpose of placing unwanted calls, can be avoided my following these suggestions:

1) Use strong Passwords: We can't stress this one enough: Use strong passwords! One of the first actions many people do when after they install their PBX, is often to create a phone extension with an easy password. Avoid using short or weak extension passwords. Please remember to use passwords of at least 8 characters, including a mix of upper and lower case along with digits. Remember to change them periodically every 2-3 months at most.

2) Public Internet: Avoid leaving your PBX systems, ATA Adapters and IP Phones open to the internet. Do not use DMZ mode on your router and do not forward ports to your equipment, unless you absolutely know what you are doing. This is only needed on specific cases, and only leave it open to the internet if you have experience on how to properly manage security on equipment that is open to the internet.

3) Asterisk Tweak: If you are using an Asterisk based PBX, add the following line to the sip.conf file under the [general] section and issue a reload
alwaysauthreject = yes

What this parameter does, is that it will always return an authentication error instead of a .404 not found:., even when the extension doesn't exist. This steps-up the difficulty for brute force scanners when they are attacking your PBX.

4) Trixbox, PBX In a Flash and other web interface based PBX: Change the default password. Different flavors of PBX installs come with default administration passwords. Make sure to change the default passwords immediately after your installation and also make sure the web interface is not reachable from the internet.

5) PBX Dial Plan: Do you make international calls? If no, do not allow international calls to be placed from your PBX. In Asterisk, remove ._011.. Or .00_. . Never use ._... If you are only calling a few countries on a regular basis, enable these countries only. For example: The only country you're calling is UK? Only configure _01144. In your dialplan.

6) Use additional caution while travelling: Do you plan on using a soft phone at a random internet cafe? Make sure you remove your login details after using it, and uninstall the software if possible.

7) Asterisk and Fail2ban: As an additional step you can install an additional security tool such as fail2ban, which is a free brute force detection system, it scans the log files of your PBX and then takes action based on the entries of those logs.
(http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk)
We also offer the optional service of installing fail2ban into your Asterisk PBX. A trained linux Asterisk professional can install it on your system for a one time fee of $150 USD.

8) Have your PBX Equipment listen to a different port than 5060: If your PBX is open to the internet, you can drastically reduce scan / brute force attempts by using a different SIP Port for incoming connections.

9) There are various other measures that you can perform to secure your VoIP equipment, however this email covers some of the most important aspects. The technology and the methods used by abusers keep evolving constantly. Meeting the recommendations on this email you will have a more secure PBX system.

PBX Security

2011-05-03T14:02:46Z

Martinm: PBX Security