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Cisco SPA112

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: '''There have been some reports of issues with this device, from both customers of VoIP.ms and other providers.'''
: '''There have been some reports of issues with this device, from both customers of VoIP.ms and other providers.'''
-
: '''Make sure to install the latest firmware from [http://software.cisco.com/download/release.html?mdfid=283998793&softwareid=282463187&release=1.3.1&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
+
: '''Make sure to install the latest firmware from [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.0&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
: '''Version 1.1 or later should be used for proper Caller ID support. '''
: '''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. '''
: '''Some People have reported issues using Firefox to Configure this device please try Chrome or IE. '''
== Configuration Details ==
== Configuration Details ==
-
'''Step 1'''
 
-
Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
+
=== Getting the IP address of your device ===
-
:Dial **** from the phone, even though there is no dial tone.  
+
 
-
: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)
+
There are two ways to retrieve the IP address of your Cisco SPA112, via analog phone menu, and via your internet router.
 +
 
 +
{| class="wikitable"
 +
! style="text-align: center;" | Analog phone interface
 +
! style="text-align: center;" | Internet Router
 +
|-
 +
|
 +
# Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
 +
## Dial **** from the phone, even though there is no dial tone.  
 +
## 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'''
  '''If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address'''
Line 23: Line 31:
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. Dialling ****110# will give one address, ****210# will give the other.
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. Dialling ****110# will give one address, ****210# will give the other.
 +
|
 +
# Attach the Cisco SPA112 to your network
 +
# Access your router's remote administration interface via your web browser, typical addresses may be <code>192.168.0.1</code> or <code>192.168.1.1</code>. Refer to your router instructions for more information.
 +
# Enter your username/password if challenged. If you have not set one, then it is likely the unchanged default password.
 +
# 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"
 +
# Navigate to this IP address via your web browser
 +
|}
-
'''Step 2'''
+
 
 +
 
 +
=== 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).
Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1).
Line 31: Line 48:
For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.
For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.
-
'''Step 3'''
+
=== Configuring the Quick Setup screen ===
-
 
+
-
The remaining steps are identical for both SPA112 and SPA122.
+
-
 
+
-
[[File:SPA112_Quick.PNG]]
+
Go to Quick Setup and configure Line 1 as follows:
Go to Quick Setup and configure Line 1 as follows:
-
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)
+
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])
'''Display Name:''' Your name
'''Display Name:''' Your name
-
'''User ID:''' Your VoIP.MS SIP Account number (or subaccount)
+
'''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
'''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|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)
+
'''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.)
+
(note: Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)
Click Submit to save settings.
Click Submit to save settings.
-
'''Step 4'''
+
[[File:quick_setup_test.png|800px|thumb|left|Quick Setup Page - click to enlarge]]
 +
<br clear="all" />
 +
 
 +
=== Configuring the Voice Line ===
 +
==== Nat Settings ====
Click on Voice, then Line 1
Click on Voice, then Line 1
Line 59: Line 76:
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.
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.
-
Under '''Proxy and Registration''' set '''Register Expires''' to 180, '''Proxy Fallback Intvl''' to 180
+
[[File:VL_1_nat_settings.png|800px|thumb|left|NAT Settings - click to enlarge]]
 +
<br clear="all" />
-
[[File:SPA112_NAT1.png]]
+
==== Proxy and Registration ====
-
[[File:SPA112_Proxy1.png]]
+
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 [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
 +
'''Register Expires''' to 300
 +
'''Proxy Fallback Intvl''' to 300
 +
Also confirm the following settings:<br>
Also confirm the following settings:<br>
 +
'''Register:''' YES <br>
'''Use DNS SRV:''' NO<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO
'''DNS SRV Auto Prefix:''' NO
 +
 +
[[File:VL_2_proxyAndRegistration.png|800px|thumb|left|Proxy and Registration - click to enlarge]]
 +
<br clear="all" />
Click Submit to submit these changes
Click Submit to submit these changes
-
'''Step 5'''
+
==== Subscriber Information ====
-
Click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''
+
In this section please confirm that you have the proper account information:
-
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.
+
'''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
-
Click Submit to save the changes
 
-
'''Step 6'''
+
[[File:VL_3_subscriberInformation.png|800px|thumb|left|Subscriber Information - click to enlarge]]
 +
<br clear="all" />
 +
 
 +
==== 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)
 +
 
 +
[[File:VL_4_audioConfig.png|800px|thumb|left|Audio configuration - click to enlarge]]
 +
<br clear="all" />
 +
 
 +
 
 +
==== 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.)
 +
 
 +
[[File:VL_5_dialPlan.png|800px|thumb|left|Dial Plan - click to enlarge]]
 +
<br clear="all" />
 +
 
 +
You can create your own dial plan if you need it, referring to this entry [[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'''.
Click '''Voice''', then go to '''SIP'''.
Set SIP Timer Values (sec)
Set SIP Timer Values (sec)
-
:SIP T1:   '''1'''
+
 
 +
    SIP T1: 1  
Set RTP Parameters
Set RTP Parameters
-
:RTP Packet Size:    '''0.02'''
 
-
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".
+
    RTP Packet Size: 0.02
 +
    RTP Port Min: 10000
 +
    RTP Port Max: 20000
-
Click Submit to save the changes
+
Click Submit to save the changes  
-
== Firmware upgrade ==
+
[[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
 +
 
 +
 +
==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'''
 +
 
 +
[[File:Pap2Ring.jpg|800px|thumb|left| Ring and Call Waiting - click to enlarge]]
 +
<br clear="all" />
 +
 
 +
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.
 +
 
 +
[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - click to enlarge]]
 +
<br clear="all" />
 +
 
 +
=== 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 the VoIP.ms (or other provider's) web interface.
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 the VoIP.ms (or other provider's) web interface.
-
Updated firmware is available from the Cisco site [http://software.cisco.com/download/release.html?mdfid=283998793&softwareid=282463187&release=1.3.1&relind=AVAILABLE&rellifecycle=&reltype=latest] 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 reinitialise, 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 reinitialisation is complete.
+
Updated firmware is available from the Cisco site [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.0&relind=AVAILABLE&rellifecycle=&reltype=latest 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.
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.
Line 104: Line 208:
[[category:Analog Telephone Adapters]]
[[category:Analog Telephone Adapters]]
 +
 +
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]
 +
 +
 +
=== SPA Star Codes ===
 +
 +
* *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 Deact 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
 +
Disables 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.)

Latest revision as of 15:20, 10 October 2016

Cisco SPA112
There have been some reports of issues with this device, from both customers of 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

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 initialise; even a correctly configured and installed SPA112/122 will give no power to the 'phone or no dialtone until initialisation 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. Dialling ****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 challenged. 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.

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 submit 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.)
Dial Plan - click to enlarge


You can create your own dial plan if you need it, referring to this entry 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.

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 the 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

This code calls the last caller.

Redials the last number called. (Not in pap2t)

Begins a blind transfer of the current call to the extension specified after the activation code.

Starts a callback when the last outbound call is not busy.

Cancels a callback.

Starts a callback when the last outbound call is busy. (Not in pap2t)

Forwards all calls to the extension specified after the activation code.

Cancels call forwarding of all calls.

Forwards busy calls to the extension specified after the activation code.

Cancels call forwarding of busy calls.

Forwards no-answer calls to the extension specified after the activation code.

Cancels call forwarding of no-answer calls.

Forwards the last inbound or outbound calls to the extension specified after the activation code.

Cancels call forwarding of the last inbound or outbound calls.

Blocks the last inbound call.

Cancels blocking of the last inbound call.

Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.

Cancels the code to accept the last outbound call.

Enables call waiting on all calls.

Disables call waiting on all calls.

Enables call waiting for the next call.

Disables call waiting for the next call.

Blocks caller ID on all outbound calls.

Removes caller ID blocking on all outbound calls.

Blocks caller ID on the next outbound call.

Removes caller ID blocking on the next inbound call.

Blocks all anonymous calls.

Removes blocking of all anonymous calls.

Enables the do not disturb feature.

Disables the do not disturb feature.

Enables caller ID generation.

Disables caller ID generation.

Enables call waiting, caller ID generation.

Disables call waiting, caller ID generation.

Enables the distinctive ringing feature.

Disables the distinctive ringing feature. The default is *46.

Assigns a speed dial number.

Makes all outbound calls secure.

Makes all outbound calls not secure.

Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)

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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