Manually configuring devices

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Please note these instructions are for advanced users, ideally all devices would be connected using auto device provisioning using instrustions available to signed in users.

In the following sections we will show you how to do this on an Aastra, Yealink T22, Cisco 504G, and Polycom 550

How to configure an Aastra 6739i

  1. Find the IP Address of your phone.  This can be done on the phone by the following: Press Options button, then Status button, then Network button, this will give you the IP
  2. Once you have your IP Address, go into your browser and input it, then log in.  The default username/password is admin/22222.
  3. Once logged in, click on Line 1 under Advanced Settings on the left hand side of screen.
  4. Enter Screen Name, Phone Number, Auth Name, Password, Proxy Server and port, Outbound Proxy Server and port, Backup Outbound Proxy Server and port, Registrar Server and port, and Registration Period as in image.
  5. Click Save Settings
  6. Your phone should now be registered  Happy calling!

 

For an Aastra the following must all be the same: Phone Number, Authentication Name, SIP username, and extension number.

How to configure a Yealink T22

  1. Find the IP Address of your phone.  This can be done on the phone by the following: Press the OK button.
  2. Once you have your IP Address, go into your browser and input it, then log in.  The default username/password is admin/admin.
  3. Once logged in, click on the Account tab
  4. Enter Select On, enter Label, Display Name, Register Name, User Name, Password, SIP Server and port, select enabled for outbound proxy, enter Outbound Proxy Server and port, select UDP, enter Backup Outbound Proxy Server and port, select disabled for NAT Traversal, enter *97 for Voice Mail, select enabled for Missed Call log if you wish, as in image.
  5. Click Confirm
  6. Now go to Advanced, enter 60 for the Login Expire(seconds) and a unique port for the Local SIP Port which is not 5060 and click Confirm.
  7. Your phone should now be registered.  Happy calling!

How to configure a Cisco 504G

 

  1. First of all, find your phone IP address. To do so, press the button that looks like a sheet of paper and then “9” (on your dialpad). You should now see it under “currentIP”.
  2. Type this IP address in your favorite web browser address bar. The page that you are landing on should look like this:
  3. You will need to do the modification as an administrator. Click on “Admin Login” on the top right corner:
  4. Then switch to the “advanced” mode (same place)
  5. Let’s first start to setup the SIP/proxy stuff which are the only mandatory settings to be able to make calls. Navigate to the “Ext 1” section:
  6. Navigate to the “Proxy and Registration” section and fill the inputs with your servers information (keep in mind that, for a weird reason, “Proxy” is actually the SIP server and the “Alternate Proxy” is the backup SIP server; for our system you do not have to enter the “Alternate Proxy). Change “Use Outbound Proxy” to yes and the “Register Expires” to something around 120:
  7. Navigate to the “Subscriber information”. The 4 inputs that you need to fill out are the display name, the password, the Auth ID and the User ID (change “Use Auth ID” to yes).  You can get these from the Device SIP settings in the KazooUI.  For Auth ID use the same as the User ID:
  8. You will need to change what’s inside of the “Dial Plan” input (“Dial Plan” section) to this: (*.|*.x.|x.) – Yeah you need to put the parentheses. Click on the “Submit all changes” at the bottom of the page and wait… You should now be able to make calls.

What you can see is that on the phone, the 4 lines are setup as the 1st one. To change that, you need to go to the “Phone” section in the navigation bar and in it go to the Line Key [1 – 2 – 3 – 4] section and set “Extension” for line 2 – 3 – 4 to “Disabled”:

You have the possibility to change the codec settings for each Line in the “Audio Configuration” section: 

If you want to change the time settings, you will need to “Regional” in the navigation bar and then “Miscellaneous” section: 

How to configure a Polycom SoundPoint IP 550

 

  1. Find the IP Address of your phone.  This can be done on the phone by the following: Press the Menu button, go to Status, press check button, go to Network, press check button, go to TCP/IP Parameters. press the check button, and the IP should appear along with other info.
  2. Once you have your IP Address, go into your browser and input it, then log in.  The default username/password is admin/456.
  3. Once logged in, hover over Settings and select Lines
  4. Expand Identification, Outbound Proxy, and Server 1
  5. Under Identification enter the Display Name, Address, Authentication User ID, Authentication Password, and Label.  You should use your Device Username and Password.
  6. Under Outbound Proxy enter the Address which is our Proxy server (See bottom of this page for details) and set Port to 7000.
  7. Under Server 1 enter the Address which is the realm for your account, set Port to 5060 and set Expires (s) to 60.
  8. Click Save.

Linksys ATA (SPA3102)

In order to configure the ATA, you will first need to get the IP address of the ATA. To do so, the easiest way is to connect directly to the ATA using the “Ethernet” port and then in your browser go to 192.168.0.1. You should then see the ATA interface and be able to get its IP address.

It should look like: 

Clic on “Admin login” in the top right corner and then on “Advanced”.

You can now clic on “Voice”. You should now see something like:

Now you must configure the “Line 1” as follow: 

Make sure to set “Use Outbound Proxy” and “Use Auth ID” to “yes”.

You can do the same thing with PSTN Line if you need to use your landline.

If you need to configure the ATA so that it handle calls on different interfaces for different numbers then you might want to do as follow:

At the bottom of the “Line 1” tab you will find a setting called “Dial Plan”. Let’s say that you want to route any call that would be like 1+ number to Kazoo and any extension (2 – 5 digits) via the PSTN (gw0) You would have a Dial Plan like:

(xx<:@gw0>|xxx<:@gw0>|xxxx<:@gw0>|xxxxx<:@gw0>|1xxxxxxxxxx)

Grandstream ATA Recommended Configuration Settings

 

 

  1. Once you obtain the IP Address go into your browser and input it, then log in.  The default password is admin.
  2. Click on the Basic Settings Tab:
  3. In the Time Zone field: Please choose one from the drop down box.
  4. Click on the Advanced Settings Tab.
  5. Use STUN to detect network connectivity:  NO
  6. Automatic Upgrade:  YES
  7. Click on the FXS PORT1 Tab.

  8. Below are other recommended Settings in the FXS Port1 tab as well.
    1. Account Active: YES
    2. Primary SIP Server : realm (information provided from portal.SIPadminhz.com under “Device”)
    3. Outbound Proxy: Use one of the SIP Proxies if using our Hosted Platform:

    sipproxy001-aa-dfw.SIPadminhz.com

    sipproxy001-aa-ord.SIPadminhz.com

    1. NAT Traversal (STUN): Keep-Alive  (must be set on both FXS Ports)
    2. SIP User ID: Username (information provided from portal.SIPadminhz.com under “Device”
    3. Authenticate ID: Username (information provided from portal.SIPadminhz.com under “Device”
    4. Name: Username (information provided from portal.SIPadminhz.com under “Device”
    5. Outgoing Call without Registration: YES
    6. Register Expiration: 1
    7. Local SIP Port: random 4 digit number
    8. Local RTP Port: a different random 4 digit number
    9. Use Random Port: NO
    10. Validate Incoming SIP message: NO
    11. Check SIP User ID for incoming INVITE: NO
    12. Allow Incoming SIP Messages for SIP Proxy ONLY: NO
    13. Preferred DTMF method:
      1.  Priority 1: RFC2833
      2. Priority  2: RFC2833
      3. Priority  3: RFC2833
    14. Disable DTMF Negotiation: NO
    15. Send Hook Flash Event: YES
    16. Enable Call Features: NO
    17. Ring Timeout: 300
    18. Use # as Dial Key: YES
    19. Subscribe for MWI: NO
    20. VAD: NO
    21. Symmetric RTP: YES
    22. Disable Line Echo Canceller (LEC): NO
    23. Remove OBP from route Header: YES