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Cisco SPA112

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You can create your own dial plan if you need it. See [[Dial Plan for Linksys ATAs]]
You can create your own dial plan if you need it. See [[Dial Plan for Linksys ATAs]]
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=== Optional settings  ===
 +
 +
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==== Outbound audio "breaking up". ====
 +
 +
Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".
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Click '''Voice''', then go to '''SIP'''.
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Set SIP Timer Values (sec)
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    SIP T1: 1
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Set RTP Parameters
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    RTP Packet Size: 0.02
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    RTP Port Min: 10000
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    RTP Port Max: 20000
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Click Submit to save the changes
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[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - Click to enlarge]]
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<br clear="all" />
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==== Caller ID display showing incorrect time ====
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Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:
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Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''
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Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.
===Configuring a Voice line using TLS===
===Configuring a Voice line using TLS===
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'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain on how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.
These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enable it yet, please follow these instructions before going further:
These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enable it yet, please follow these instructions before going further:
-
[[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]
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For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]
====Verifying the device's Firmware version====
====Verifying the device's Firmware version====
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[[File:SPA_Regional_Tone.png|800px|thumb|left|Click to enlarge]]
[[File:SPA_Regional_Tone.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />
<br clear="all" />
-
 
-
=== Optional settings  ===
 
-
 
-
 
-
==== Outbound audio "breaking up". ====
 
-
 
-
Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".
 
-
 
-
Click '''Voice''', then go to '''SIP'''.
 
-
 
-
Set SIP Timer Values (sec)
 
-
 
-
    SIP T1: 1
 
-
 
-
Set RTP Parameters
 
-
 
-
    RTP Packet Size: 0.02
 
-
    RTP Port Min: 10000
 
-
    RTP Port Max: 20000
 
-
 
-
Click Submit to save the changes
 
-
 
-
[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - Click to enlarge]]
 
-
<br clear="all" />
 
-
 
-
==== Caller ID display showing incorrect time ====
 
-
 
-
Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:
 
-
 
-
Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''
 
-
 
-
Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.
 
Click Submit to save the changes  
Click Submit to save the changes  

Revision as of 15:21, 29 August 2019

Cisco SPA112
There have been some reports of issues with this device from customers of both VoIP.ms and other providers.
Make sure to install the latest firmware from Cisco Software.
Version 1.1 or later should be used for proper Caller ID support.
Some People have reported issues using Firefox to Configure this device; please try Chrome or IE.

Contents

Documentation

Official Configuration guide / User guide / Data Sheets

Configuration Details

Getting the IP address of your device

There are two ways to retrieve the IP address of your Cisco SPA112: via analog phone menu, and via your internet router.

Analog phone interface Internet Router
  1. Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
    1. Dial **** from the phone, even though there is no dial tone.
    2. When you hear "System Configuration Menu," dial 1 1 0 # slowly. The current IP address will be read back. (e.g. 192.168.X.X)
If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address
can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad
(for example, 10*1*27*2 for 10.1.27.2). The network mask can be set with option 121# and the default
gateway can be sent with option 131#
Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/DOC-9900 document.

Be sure to allow at least a minute or two for the box to initialize; even a correctly configured and installed SPA112/122 will give no power or dialtone to the phone until initialization is complete.

Note that the SPA122 is basically a SPA112 with a second network port, intended for installation between a local network hub (LAN) and an upstream Internet (WAN) connection. The SPA122 may be configured as either a "NAT" or "bridge". Depending on configuration, this leaves the SPA122 with two addresses; a local area network address (such as 192.168.15.1) and an outside Internet address. Dialing ****110# will give one address, ****210# will give the other.

  1. Attach the Cisco SPA112 to your network
  2. Access your router's remote administration interface via your web browser (typical addresses may be 192.168.0.1 or 192.168.1.1). Refer to your router instructions for more information.
  3. Enter your username/password if asked. If you have not set one, then it is likely the unchanged default password.
  4. In the router's menu, there should be a page showing a list of connected clients, with their internal IP address. Find the entry corresponding to the Cisco SPA112. It should identify itself in the list as "SPA112"
  5. Navigate to this IP address via your web browser


Accessing to the device's settings page

Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1). The default username is admin, and the default password is also admin.

For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.

Configuring the Quick Setup screen

Go to Quick Setup and configure Line 1 as follows:

Proxy: atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)

Display Name: Your name

User ID: 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

Password: Your VoIP.MS SIP Password

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.)

Click Submit to save settings.

Quick Setup Page - Click to enlarge


Configuring the Voice Line

Nat Settings

Click on Voice, then Line 1

Set NAT Mapping Enable to Yes, then set NAT Keep Alive Enable to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its traffic will not be subject to NAT in this configuration.

If using the second phone line on an SPA122 device, change the SIP Port for one of the lines to e.g. 5080.

NAT Settings - Click to enlarge


Proxy and Registration

Under Proxy and Registration, set the server you will use as registration server and the proper values for the Register Expires and Proxy Fallback Intvl:

Proxy: atlanta.voip.ms (one of VoIP.ms multiple servers, you can choose the one closest to your location)
Register Expires to 300
Proxy Fallback Intvl to 300

Also confirm the following settings:
Register: YES
Use DNS SRV: NO
DNS SRV Auto Prefix: NO

Proxy and Registration - Click to enlarge


Click Submit to save these changes

Subscriber Information

In this section please confirm that you have the proper account information:

Display Name: Your name (that will be shown as callerID name)
User ID: 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
Password: Your VoIP.ms SIP Password


Subscriber Information - Click to enlarge


Audio Configuration

You can verify or change the audio codec that will be used with the calls. Please verify that you have the same codec selected in your SIP account's settings.

Preferred codec: g711u (or G729)

Audio configuration - Click to enlarge



Dial Plan

We recommend to use this 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.)


Dial Plan - Click to enlarge


You can create your own dial plan if you need it. See Dial Plan for Linksys ATAs


Optional settings

Outbound audio "breaking up".

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click Voice, then go to SIP.

Set SIP Timer Values (sec)

   SIP T1: 1 

Set RTP Parameters

   RTP Packet Size: 0.02 
   RTP Port Min: 10000 
   RTP Port Max: 20000 

Click Submit to save the changes

SIP Values - Click to enlarge


Caller ID display showing incorrect time

Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:

Enter your device's settings and click Network Setup, then go to Basic Setup, then click Time Settings

Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.

Configuring a Voice line using TLS

NOTE: This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain on how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enable it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on Main account or more information on how to enable encrypted traffic for the sub account sub account

Verifying the device's Firmware version

First, check your firmware's version. This is, from your Device's Configuration Utility at Status >> System Information

We strongly recommend to use the latest firmware version available, up today is the 1.4.1 (SR3) Apr 3 2019. If you do not have this version, you may consider its upgrade.
Click to enlarge


Enabling TLS for the line

Go to the User's line you will use (If you use the Line 1 go to User 1) and navigate to the Supplementary Service Settings, there set:

Secure Call Setting: yes
Click to enlarge


Configuring the transport and port

Go to the line you will be using with TLS and navigate to the section SIP Settings, then set:

SIP Transport : TLS
SIP Port : 5061
Click to enlarge


CA Certificate

As per CISCO's requirements, a CA certificate is needed to use Secure calls with the SPA112's device. To achieve this you will need to import the CA Cert.

Go to Voice >> Provisioning and once there navigate to CA Settings, at Custom CA URL enter the following:

http://spa1xx.voip.ms/cca.pem
Click to enlarge



Click Submit, the device will reboot and after that will register and you will be ready to use it with TLS

Secure Call Indication Tone

Once the secure call feature is enabled, during all the duration of your calls you will hear a couple of tones (beeps), this is normal and beyond VoIP.ms control, however you can disable this notification going to: Voice >> Regional >> Secure Call Indication Tone

Note: Please notice that this setting is not an on/off, you will need to remove all the line
In case you need to set it back, the default value is 
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
Click to enlarge


Click Submit to save the changes


Known Issues

Phone will not ring on handset

Sometimes the Phone you are using is designed for a certain voltage and ring waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings, please follow these steps to hopefully resolve this issue for you.

Step 1: First access the SPA web interface.

Step 2: Click on the Admin Login and then click on (switch to advanced view)

Step 3: Click on your Regional tab on the top menu.

Step 4: Go halfway down the page until you see the heading Ring and Call Waiting Tone Spec

Ring and Call Waiting - Click to enlarge


Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save settings and test an incoming call.

Receiving Unwanted Calls in the middle of the Night (i.e. CallerID 100) that do not appear in your CDR:

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

Restrict IP - Click to enlarge


Firmware Upgrade

SPA112 and SPA122 adapters were distributed with outdated (1.0.x) firmware at least as late as 2012; affected boxes will not show Caller ID on any inbound call, even though the caller names and numbers are visible in the call detail record on VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site Cisco Firmware as a .ZIP archive which contains two files (a .BIN with the actual firmware and a .PDF with documentation). Download and unzip this file. Go to the 'administration' tab on the web interface (on the SPA122, this needs to be done from the LAN side with SPA122's built-in networking set to NAT mode). On the left sidebar, click 'update firmware' (as most of the administration menu does not appear for Firefox users, downgrade to MS IE or another browser temporarily). Click the 'upload' button and indicate the location of the unzipped .BIN file. A box will appear with a progress indicator and a warning not to interrupt the upgrade. When the upgrade is completed, the SPA112/122 will reset and will likely take a minute or more to reinitialize, reconnect to the network and restore dial tone. SPA122 users who have installed the device in-line between the local PCs and the Internet will be disconnected from the Internet until reinitialization is complete.

Once the new firmware is deployed, call display will operate normally and the configuration web page will display in Firefox without missing options in the administration menu.

A manual for Cisco's SPA100 series adapters is online at http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag.html


SPA Star Codes

SPA Star Codes
Star Code Name Result
*69 Call Return Code This code calls the last caller.
*07 Call Redial Code Redials the last number called. (Not in pap2t)
*98 Blind Transfer Code Begins a blind transfer of the current call to the extension specified after the activation code.
*66 Call Back Act Code Starts a callback when the last outbound call is not busy.
*86 Call Back Dea t Code Cancels a callback.
*05 Call Back Busy Act Code Starts a callback when the last outbound call is busy. (Not in pap2t)
*72 Cfwd All Act Code Forwards all calls to the extension specified after the activation code.
*73 Cfwd All Deact Code Cancels call forwarding of all calls.
*90 Cfwd Busy Act Code Forwards busy calls to the extension specified after the activation code.
*91 Cfwd Busy Deact Code Cancels call forwarding of busy calls.
*92 Cfwd No Ans Act Code Forwards no-answer calls to the extension specified after the activation code.
*93 Cfwd No Ans Deact Code Cancels call forwarding of no-answer calls.
*63 Cfwd Last Act Code Forwards the last inbound or outbound calls to the extension specified after the activation code.
*83 Cfwd Last Deact Code Cancels call forwarding of the last inbound or outbound calls.
*60 Block Last Act Code Blocks the last inbound call.
*80 Block Last Deact Code Cancels blocking of the last inbound call.
*64 Accept Last Act Code Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
*84 Accept Last Deact Code Cancels the code to accept the last outbound call.
*56 CW Act Code Enables call waiting on all calls.
*57 CW Deact Code Disables call waiting on all calls.
*71 CW Per Call Act Code Enables call waiting for the next call.
*70 CW Per Call Deact Code Disables call waiting for the next call.
*67 Block CID Act Code Blocks caller ID on all outbound calls.
*68 Block CID Deact Code Removes caller ID blocking on all outbound calls.
*81 Block CID Per Call Act Code Blocks caller ID on the next outbound call.
*82 Block CID Per Call Deact Code Removes caller ID blocking on the next inbound call.
*77 Block ANC Act Code Blocks all anonymous calls.
*87 Block ANC Deact Code Removes blocking of all anonymous calls.
*78 DND Act Code Enables the do not disturb feature.
*79 DND Deact Code Disables the do not disturb feature.
*65 CID Act Code Enables caller ID generation.
*85 CID Deact Code Disables caller ID generation.
*25 CWCID Act Code Enables call waiting, caller ID generation.
*45 CWCID Deact Code Disables call waiting, caller ID generation.
*26 Dist Ring Act Code Enables the distinctive ringing feature.
*46 Dist Ring Deact Code Enables the distinctive ringing feature. The default is *46 .
*74 Speed Dial Act Code Assigns a speed dial number.
*16 Secure All Call Act Code Makes all outbound calls secure.
*17 Secure No Call Act Code Makes all outbound calls not secure.
*18 Secure One Call Act Code Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
*19 Secure One Call Deact Code Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
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