Cisco IP Phone 68XX and 88XX
From VoIP.ms Wiki
- The Cisco IP Phone 8800 Series is a great fit for businesses of all sizes seeking secure, high-quality, full-featured VoIP. Select models provide affordable entry to HD video and support for highly-active, in-campus mobile workers. This advanced series provides flexible deployment options for Cisco pre-approved third-party UCaaS providers
- The Cisco IP Phone 6800 Series multiplatform phones are designed for affordability. They deliver reliable, business-grade audio, with Gigabit Ethernet integration and low power usage. Ideal for customers with moderate to active VoIP needs, the 6800 Series phones are supported on Cisco-approved third-party unified communications as a service (UCaaS) providers.
While the screenshots used for this guide were taken from a Cisco's IP Phone 8865, the steps and configurations have been tested and apply for the Cisco's models:
- Cisco 8865
- Cisco 8861
- Cisco 6821
- Cisco 6851
And most likely will work with most of the Cisco's IP phones
Contents |
Configuring the IP Phone's
Getting the IP address
In order to configure the Cisco IP phone to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.
To perform this click on the "Menu" button of the phone. Once on the settings menu, navigate to:
Status >> Network Status >> IPv4 Status
On this section, you'll need to take note of the IP address shown beside: " IP address ". It should read back something as: " 192.168.0.1 ".
Loggin into the Web interface
Once you have the IP address please open a web browser of your preference and it's URL bar enter the IP address you got by prepending: " http:// " and access it. If you selected Skip on the user/password prompt when you first plugged your IP phone, you will not be asked for either of them
When the page has loaded, on the top-right corner, click Admin Login, after that click advanced, the URL will look like: http://{IP}/admin/advanced
I.E: http://192.168.100.35/admin/advanced
Configuring an extension
Go to Ext X where X is the extension you are going to use with yout SIP account.
General & NAT Settings
- Line Enable:: Yes
- NAT Mapping Enable: Yes
- NAT Keep Alive Enable: Yes
SIP Settings
- SIP Transport: UDP or TCP
- SIP Port: 5060 (5080 or 42872 are also possible)
Proxy and Registration
- Proxy: dallas1.voip.ms One of VoIP.ms multiple servers, you can choose the one closest to your location
- Outbound Proxy: dallas1.voip.ms (The same you used on Proxy)
- Register: Yes
- Register Expires: 300
Subscriber Information
- Display Name: Your CallerID Name
- User ID: 100000 (replace with your SIP main account or subaccount)
- Password: Your SIP account's password
- Auth ID: The same you used at User ID
Audio Configuration
- Preferred Codec : G729a or G711u (depending on your SIP account's settings)
- Use Pref Codec Only : Yes
- G711u Enable: Yes
- G711a Enable: No
- G729a Enable: Yes
- G722 Enable: No
- G722.2 Enable: No
- iLBC Enable: No
- iSAC Enable: No
- OPUS Enable: No
Video Configuration
If your device has video capabilities, since we do not offer Video services, please set all the options to "No"
- H264 BP0 Enable:: No
- H264 BP1 Enable:: No
- H264 HP Enable:: No
Dial plan
(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)
Note: Replace 555 in the dial plan with your area code, See Dial Plan for Linksys ATAs for details.
You can create your own dial plan if you need it. See Dial Plan for Linksys ATAs and also here you'll find a great explanation about dial plans for Cisco devices
Once all the configurations needed have been done, do not forget to scroll to the bottom of the page and click on Submit All Changes, this will save, apply and reload the IP Phone's settings. You'll see a Phone is updating configuration message, do not unplug the phone during this state. After a few seconds, the device will be ready to be used.
TLS
Extension's settings
SIP Settings
- SIP Transport: TLS
- SIP Port: 5061 (5081 or 42873 are also possible)
Audio Configuration
Verify that the following value is set properly
- Encryption Method: AES128
User Tab's setting
It is needed to enable "Secure Calls" in order to have Outgoing calls working. To do so, navigate to:
Voice >> User Tab, there at the section Supplementary Services set:
- Secure Call Setting: Yes
Provisioning's Tab
In some devices, a certificate is needed in order to have them registering while using TLS, as is the case for the Cisco's IP phone 6821. If your device needs it, Go to:
Voice >> Provisioning >> CA Settings and set:
- Custom CA Rule: http://spa1xx.voip.ms/cca.pem