{{tag>Setup Cisco Cisco_7800_Series Cisco_8800_Series Configuration}}
====== How to Set Up the Cisco IP Phone 7800 and Cisco IP Phone 8800 Series IP Phones ======
This guide explains how to set up the **Cisco IP Phone 7800** series and **Cisco IP Phone 8800** series IP phones.

This guide will apply for the following handsets:
  * Cisco IP Phone 7811
  * Cisco IP Phone 7821
  * Cisco IP Phone 7841
  * Cisco IP Phone 7811
  * Cisco IP Phone 8861
  * Cisco IP Phone 8831
  * Cisco IP Phone 8832
  * Cisco IP Phone 8845
  * Cisco IP Phone 8851
  * Cisco IP Phone 8861
  * Cisco IP Phone 8865


For more information on the Cisco range of IP phones, please visit the [[https://www.cisco.com/c/en/us/products/collaboration-endpoints/index.html|Cisco phones product page]].

===== Configuration =====
  - Obtain the handset IP address.\\ //Press the page (Setup) button, scroll down to 9. Network and note the CurrentIP//
  - Type the IP address into your web browser.
  - Once the page has loaded, press **Admin Login** then press **advanced**
  - Go to the **SIP** tab
  - Scroll down to **RTP Parameters** and sets **RTP Packet Size** to **0.020** :!://This is often overlooked.//
  - Scroll down to **NAT Support Parameters** and set **Handle VIA rport** and **Insert VIA rport** to **no**
  - Go to the **Ext 1** tab (Or whichever Ext you are using)
    * Under **General** set **Line Enable** to **yes**
    * Under **NAT Settings** set **NAT Mapping** and **NAT Keep Alive** to **yes**
    * Under **SIP Settings** set **SIP Transport** to **UDP**
    * Under **Proxy and Registration** enter the following settings. You will need to reference your welcome pack.
      * Set **Proxy** to the //SIP Proxy/Registrar Server// provided in your welcome email
      * Set **Register** to **yes**
    * Under **Subscriber Information** enter the following settings. You will need to reference your welcome pack.
      * Put your name into **Display Name**. What is entered here is arbitrary.
      * Put your //SIP Username (extension number)// into the **User ID** and **Auth ID** fields. This is provided in your welcome pack.
      * Put your provided //SIP password (also referred to as SIP Secret)// into the **Password** field. This is provided in your welcome pack.
      * Set **Use Auth ID** to **yes**
    * Under **Audio Configuration** set **Preferred Codec** to **G711a** or set to **G722** if on SureVoIP Multi-User Hosted.
    * Put the following into **Dial Plan**\\ <code>([1234]xxx|0[1-9]xxxxxxxxxS0|0[1-9]xxxxxxxx|*09x|*9[78]S0|**[1234]xxxS0|*99[1234]xxxS0|999S0|112|101|111|116xxx|00xxx.|xxxxxxx.)</code>
  - Click **Submit All Changes** 
  - Once the phone has rebooted and is back online, go to the **Info**  tab and scroll down to **Ext 1 Status** and check the **Registration State** shows **Registered**

===== Feature Keys =====
==== Line Keys ====
The line keys, along the display, can be programmed to allow you to select which line you wish to use to make a call (if you use your phone with more than one account).

Under the **Phone** tab, select which extension each line key should correspond to. You can give each a short name that will appear alongside the button on the display.

Or if you have buttons spare, they can be set up to be BLF + speed dial + call pickup buttons.

This allows you to monitor the state of another extension (if it is ringing, busy or available), quickly dial that extension, or pick up an extension that is ringing (and there is nobody there).

  - First, click the **advanced** button on the top-right.
  - Go to the **Attendant Console** tab
    * Set **Server Type** to **Asterisk**
    * Set **Attendant Console Call Pickup Code** to **%%**#%%** 
  - Go to the **Phone** tab
    * For each button you wish to use as a BLF+SD+CP key, set **Extension** to **Disabled**
    * Type in something short yet relevant into the **Short Name** field, such as the person's name
    * In **Extended Function** put in: **fnc=sd+blf+cp;sub=XXXX@$PROXY**\\ //**XXXX**// is the extension you wish to monitor
    * Alternatively, for Speed Dial only to an external number, simply put in: **fnc=sd;ext=XXXXXXXXXXX@$PROXY**\\ Where XXXXXXXXXXX is the number to dial.
  -  When the monitored extension is available, it shows green. Red when on a call or busy. Blinking red when ringing. Orange if there is an error in the syntax.  
  - It is highly recommended to remove any conflicting **Vertical Service Activation Codes** in the **Regional** tab. Simply clear them out and click **Submit All Changes**. 
  - **Please note that at least 1 line key is required for the actual line, otherwise your phone will be unable to function properly.**

==== Voicemail ====
To program the **Voicemail** key to dial your voicemail, go to the **Phone** tab, and in the **Voice Mail Number** field put in ***97**


==== Additional Settings ====
These are optional settings, but are highly recommended to get the best experience out of your phone.

=== Set Time Servers ===
In **System** tab, under **Optional Network Configuration** 
  * Set **NTP Enable** to **//yes//**
  * Set **Primary NTP Server** to **//0.uk.pool.ntp.org//** 
  * Set **Secondary NTP Server** to **//1.uk.pool.ntp.org//**
//Use the above as an example as your system administrator may be able to provide you with your company's internal NTP servers.//

In **Regional** tab, under **Miscellaneous**
  * Set **Time Zone** to **//GMT//**
  * Set **Daylight Savings Rule** to **start=3/-1/7/2:0:0;end=10/-1/7/2:0:0;save=1:0:0**
  * Set **Ignore DHCP Time Offset** to **//no//**
  * Set **Locale** to **//en-GB//**


===== Troubleshooting =====
If registration is failing, check over the settings entered above.

Ensure [[troubleshooting:sip_alg|SIP ALG]] is disabled on your router. See [[troubleshooting:sip_alg|this page]] for further information.

If the Status page shows timeout, check over your [[..:..:nat_and_firewall_settings|firewall settings]].

If you are still having trouble registering, contact [[http://www.surevoip.co.uk/contact-surevoip|SureVoIP Technical Support]].