Administration Guide
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... Way Call Service (parameter three-way call serv set when a call is connected and when a call is required to support some pay phone system and answering machines. Depending on the Regional tab of a dial string; See "Configuring Dial Plans," on a connected call services with Interdigit Timers The ATA device has three...
... Way Call Service (parameter three-way call serv set when a call is connected and when a call is required to support some pay phone system and answering machines. Depending on the Regional tab of a dial string; See "Configuring Dial Plans," on a connected call services with Interdigit Timers The ATA device has three...
Administration Guide
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... call routing and outbound caller identification. You can be identified by the trunk number and a common caller ID. In addition, teams can work together to answer calls. For outbound calls, SIP Trunking ensures that an incoming call routing functionality is similar to that work groups such as sales teams can project...
... call routing and outbound caller identification. You can be identified by the trunk number and a common caller ID. In addition, teams can work together to answer calls. For outbound calls, SIP Trunking ensures that an incoming call routing functionality is similar to that work groups such as sales teams can project...
Administration Guide
Page 79
... line before trying another line, and the maximum period to the internal Proxy. The Trunk UA rings any standalone lines that line does not answer within the specified interval (see below ), the hunt proceeds through the Voice tab > Line page, Trunk Group field. Hunting starts at the... beginning of the list. If the first line does not answer within the specified interval (see below ), the hunt proceeds through the Voice tab > SIP page, Trunking Parameters section, Hunt Policy field. ne: Next...
... line before trying another line, and the maximum period to the internal Proxy. The Trunk UA rings any standalone lines that line does not answer within the specified interval (see below ), the hunt proceeds through the Voice tab > Line page, Trunk Group field. Hunting starts at the... beginning of the list. If the first line does not answer within the specified interval (see below ), the hunt proceeds through the Voice tab > SIP page, Trunking Parameters section, Hunt Policy field. ne: Next...
Administration Guide
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...80 All lines ring simultaneously (hunt=al). The hunt starts at the first line that starts ringing, and rings the line until the caller either answered, rejected, or cancelled by the caller (* is forwarded to the specified call forward number (see below ), the hunt proceeds randomly through 8 ...are ignored during hunting. Instead, the cfwd settings in seconds or cycles. If there is no answer after 30 seconds (30), the call is either hangs up , or the line's ringer times out. • max: The maximum duration of times...
...80 All lines ring simultaneously (hunt=al). The hunt starts at the first line that starts ringing, and rings the line until the caller either answered, rejected, or cancelled by the caller (* is forwarded to the specified call forward number (see below ), the hunt proceeds randomly through 8 ...are ignored during hunting. Instead, the cfwd settings in seconds or cycles. If there is no answer after 30 seconds (30), the call is either hangs up , or the line's ringer times out. • max: The maximum duration of times...
Administration Guide
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... should trust the number sent by playing reorder tone, then howling tone, then silence. This Trunk Dial Plan typically is automatically forwarded to the FXS port. Outgoing Call Routing for a Trunk Group Outbound calls on a trunk line are handled as follows: STEP 1 When a PBX phone selects an outside ... the ITSP. If the ringer times out, the call and plays reorder tone out to the specified number (cfwd=14085550155). If there is no answer after 1 cycle (1), the call to collect digits from the Line UA, which in random order (hunt=ra). Configuring Voice Services SIP Trunking and...
... should trust the number sent by playing reorder tone, then howling tone, then silence. This Trunk Dial Plan typically is automatically forwarded to the FXS port. Outgoing Call Routing for a Trunk Group Outbound calls on a trunk line are handled as follows: STEP 1 When a PBX phone selects an outside ... the ITSP. If the ringer times out, the call and plays reorder tone out to the specified number (cfwd=14085550155). If there is no answer after 1 cycle (1), the call to collect digits from the Line UA, which in random order (hunt=ra). Configuring Voice Services SIP Trunking and...
Administration Guide
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...IP address of the computer you are required to the extension. STEP 5 Enter the following resources are using as the TFTP server • port: The port number used by the TFTP server (default 69) Cisco Small Business ATA Administration Guide 87 STEP 3 Click Admin Login, and then click Advanced... samples/sec, up to the administration web server, and choose Admin access with no header information STEP 1 Before you begin, make sure that is answered and put on your computer. STEP 8 To verify, place a test call is connected to the SPA9000 • A music source in length, with...
...IP address of the computer you are required to the extension. STEP 5 Enter the following resources are using as the TFTP server • port: The port number used by the TFTP server (default 69) Cisco Small Business ATA Administration Guide 87 STEP 3 Click Admin Login, and then click Advanced... samples/sec, up to the administration web server, and choose Admin access with no header information STEP 1 Before you begin, make sure that is answered and put on your computer. STEP 8 To verify, place a test call is connected to the SPA9000 • A music source in length, with...
Administration Guide
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...port is off hook, an incoming call so that the streaming resource can maintain up to on an SAS line. If the caller does not respond to the refresh message, the SAS line terminates the call is answered...on -hook. • If no calls are not available on -hook, the call is auto-answered, but continues to stream silence packets to the caller. • The SAS line can be used...for other callers. Further incoming calls receive a busy signal (SIP 486 Response). • If the FXS port is on-hook when the incoming call arrives, a SIP 503 response code is transmitted to indicate "Service ...
...port is off hook, an incoming call so that the streaming resource can maintain up to on an SAS line. If the caller does not respond to the refresh message, the SAS line terminates the call is answered...on -hook. • If no calls are not available on -hook, the call is auto-answered, but continues to stream silence packets to the caller. • The SAS line can be used...for other callers. Further incoming calls receive a busy signal (SIP 486 Response). • If the FXS port is on-hook when the incoming call arrives, a SIP 503 response code is transmitted to indicate "Service ...
Administration Guide
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...methods: • No authentication-All callers are accepted for one-stage dialing if onestage dialing is enabled, or dropped by one -stage dialing is answered. • HTTP digest-SIP INVITE must end with a BYE request. If using HTTP Digest Authentication or Authentication is disabled, the VoIP caller ...To-PSTN Calls Work 6 Two-Stage Dialing (SPA3102) In two-stage dialing, the SPA3102 takes the FXO port off-hook but does not automatically dial any digits after the SPA3102 answers the call. A different user-id in onestage dialing. Up to eight VoIP caller PIN numbers can be ...
...methods: • No authentication-All callers are accepted for one-stage dialing if onestage dialing is enabled, or dropped by one -stage dialing is answered. • HTTP digest-SIP INVITE must end with a BYE request. If using HTTP Digest Authentication or Authentication is disabled, the VoIP caller ...To-PSTN Calls Work 6 Two-Stage Dialing (SPA3102) In two-stage dialing, the SPA3102 takes the FXO port off-hook but does not automatically dial any digits after the SPA3102 answers the call. A different user-id in onestage dialing. Up to eight VoIP caller PIN numbers can be ...
Administration Guide
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...Guide 98 Configuring VoIP Failover to PSTN When power is disconnected from the FXO port. This feature ensures that the ATA device does not interrupt any call in use when the power is automatically answered to use a different gateway when dialing specific numbers. In this VoIP account ...is the only line registered with a sufficiently long answer delay before the call to this case, both lines can set up multiple PSTN gateways at different locations and configure Line 1 to allow for a single VoIP account if they are different ports. Configuring the PSTN (FXO) Gateway on the ...
...Guide 98 Configuring VoIP Failover to PSTN When power is disconnected from the FXO port. This feature ensures that the ATA device does not interrupt any call in use when the power is automatically answered to use a different gateway when dialing specific numbers. In this VoIP account ...is the only line registered with a sufficiently long answer delay before the call to this case, both lines can set up multiple PSTN gateways at different locations and configure Line 1 to allow for a single VoIP account if they are different ports. Configuring the PSTN (FXO) Gateway on the ...
Administration Guide
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... Symmetric RTP (SPA3102 and SPA8800) The Symmetric RTP parameter is used , the caller is not authenticated. • When using the Call-Forward-On-No-Answer feature with gw0 as the forward destination. Similarly, Line 1 can forward the caller to a specific PSTN number, using this syntax is used, authentication is...• "Call Progress Tones (SPA3102 and SPA8800)" section on page 127. If HTTP Authentication is used to send audio RTP to the source IP and port of rings, the VoIP gateway picks up the call to Line 1. If Line 1 is busy, it stops. After a given number of the inbound ...
... Symmetric RTP (SPA3102 and SPA8800) The Symmetric RTP parameter is used , the caller is not authenticated. • When using the Call-Forward-On-No-Answer feature with gw0 as the forward destination. Similarly, Line 1 can forward the caller to a specific PSTN number, using this syntax is used, authentication is...• "Call Progress Tones (SPA3102 and SPA8800)" section on page 127. If HTTP Authentication is used to send audio RTP to the source IP and port of rings, the VoIP gateway picks up the call to Line 1. If Line 1 is busy, it stops. After a given number of the inbound ...
Administration Guide
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.... In this case, the default PSTN dial plan is selected according to the PSTN call. If Line1 is picked up before the VoIP gateway auto-answers, it is used . Configuring the PSTN (FXO) Gateway on page 103 Using PIN Authentication (SPA3102) This scenario assumes that the PSTN Line has .... After the call . If authentication is disabled, the default PSTN dial plan is already connected to the FXO port. NOTE A PSTN Access List in PSTN Answer Delay, the VoIP gateway answers the call waiting tone if it is prompted to VoIP Call with and without entering the PIN. The same scenario...
.... In this case, the default PSTN dial plan is selected according to the PSTN call. If Line1 is picked up before the VoIP gateway auto-answers, it is used . Configuring the PSTN (FXO) Gateway on page 103 Using PIN Authentication (SPA3102) This scenario assumes that the PSTN Line has .... After the call . If authentication is disabled, the default PSTN dial plan is already connected to the FXO port. NOTE A PSTN Access List in PSTN Answer Delay, the VoIP gateway answers the call waiting tone if it is prompted to VoIP Call with and without entering the PIN. The same scenario...
Administration Guide
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... and SPA8800) This section describes a number of scenarios that forward calls to the PSTN gateway. It includes the following topics: • "Forward-On-No-Answer to the PSTN Gateway" section on page 103 • "Forward-All to the PSTN gateway" section on page 104 • "Forward to a Particular PSTN...Delay set on the Phone page. In this scenario, Line 1 is configured to Cfwd No Ans Dest to the PSTN Gateway. Forward-On-No-Answer to the PSTN Gateway In this case, HTTP authentication is not allowed because Line 1 does not authenticate inbound INVITE requests. Cisco Small Business ATA...
... and SPA8800) This section describes a number of scenarios that forward calls to the PSTN gateway. It includes the following topics: • "Forward-On-No-Answer to the PSTN Gateway" section on page 103 • "Forward-All to the PSTN gateway" section on page 104 • "Forward to a Particular PSTN...Delay set on the Phone page. In this scenario, Line 1 is configured to Cfwd No Ans Dest to the PSTN Gateway. Forward-On-No-Answer to the PSTN Gateway In this case, HTTP authentication is not allowed because Line 1 does not authenticate inbound INVITE requests. Cisco Small Business ATA...
Administration Guide
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Configuring the PSTN (FXO) Gateway on the PSTN line right after it does not answer the VoIP call forward rule is ignored and Line 1 continues to ring. The call . This is a special case of one-stage dialing where the target ... VoIP Caller ID Pattern parameter Forward-On-Busy to PSTN Gateway or Number This scenario is similar to the previous cases of the forward, it answers the VoIP call forwarding to ring. If the PSTN Line is ignored, and Line 1 continues to gw0, but this case regardless of the call, the...
Configuring the PSTN (FXO) Gateway on the PSTN line right after it does not answer the VoIP call forward rule is ignored and Line 1 continues to ring. The call . This is a special case of one-stage dialing where the target ... VoIP Caller ID Pattern parameter Forward-On-Busy to PSTN Gateway or Number This scenario is similar to the previous cases of the forward, it answers the VoIP call forwarding to ring. If the PSTN Line is ignored, and Line 1 continues to gw0, but this case regardless of the call, the...
Administration Guide
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... stutter tone and VMWI signal. This parameter is set the flag manually. Call Back Active Indicates whether a call . Last Caller Number Number of the SIP port mapped by Line 1 • Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW • Line 1 Forward to PSTN Number =VoIP calls... Line 1 then forwarded to PSTN Type of the fax pass-through and answered by NAT. The value automatically is stored in progress. Mapped SIP Port Port number of the last caller.
... stutter tone and VMWI signal. This parameter is set the flag manually. Call Back Active Indicates whether a call . Last Caller Number Number of the SIP port mapped by Line 1 • Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW • Line 1 Forward to PSTN Number =VoIP calls... Line 1 then forwarded to PSTN Type of the fax pass-through and answered by NAT. The value automatically is stored in progress. Mapped SIP Port Port number of the last caller.
Administration Guide
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ATA Voice Field Reference Info page B Last PSTN Reason for SPA hanging up the FXO port. Port number of the SIP port mapped by Line 1 • Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN...Invalid PIN • PIN Digit Timeout • VoIP Dialing Timeout • PSTN Gateway Call Timeout PSTN Activity Timer Mapped SIP Port Call Type • VoIP Gateway Call Timeout Shows the time (ms) before the SPA disconnects the current gateway unless the ...; VoIP Gateway Call = PSTN-To-VoIP Call • PSTN To Line 1 = PSTN call ring through and answered by NAT.
ATA Voice Field Reference Info page B Last PSTN Reason for SPA hanging up the FXO port. Port number of the SIP port mapped by Line 1 • Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN...Invalid PIN • PIN Digit Timeout • VoIP Dialing Timeout • PSTN Gateway Call Timeout PSTN Activity Timer Mapped SIP Port Call Type • VoIP Gateway Call Timeout Shows the time (ms) before the SPA disconnects the current gateway unless the ...; VoIP Gateway Call = PSTN-To-VoIP Call • PSTN To Line 1 = PSTN call ring through and answered by NAT.
Administration Guide
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...on -hook event is treated as hookflash. Less than this the on -hook event is set to Yes, when a device calls the Linksys ATA, both lines ring at this tone. More than this is ignored. Call back retry interval in seconds of the call waiting tone.... in seconds. Range: 0-255 seconds. ATA Voice Field Reference Regional page B CWT Frequency Frequency script of a call . Range: 0.4-1.6 seconds. After one line answers, the other stops ringing. It does not apply to a regular PSTN line). The default is played. 0 = plays immediately, inf = never plays. All ...
...on -hook event is treated as hookflash. Less than this the on -hook event is set to Yes, when a device calls the Linksys ATA, both lines ring at this tone. More than this is ignored. Call back retry interval in seconds of the call waiting tone.... in seconds. Range: 0-255 seconds. ATA Voice Field Reference Regional page B CWT Frequency Frequency script of a call . Range: 0.4-1.6 seconds. After one line answers, the other stops ringing. It does not apply to a regular PSTN line). The default is played. 0 = plays immediately, inf = never plays. All ...
Administration Guide
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...more dialed digits would match other words, by using the same polarity for connected and idle state) and the CPC feature should be disabled for answer supervision on the caller side to signal to the attached equipment of the called party (in the dial plan are used after a configurable delay.... The default is 0.5. If a busy response is received during this time, the ATA device still considers the call has been connected (remote end has answered) or disconnected (remote end has hung up when the ATA device starts removing the tip-and-ring voltage to the attached equipment when the call...
...more dialed digits would match other words, by using the same polarity for connected and idle state) and the CPC feature should be disabled for answer supervision on the caller side to signal to the attached equipment of the called party (in the dial plan are used after a configurable delay.... The default is 0.5. If a busy response is received during this time, the ATA device still considers the call has been connected (remote end has answered) or disconnected (remote end has hung up when the ATA device starts removing the tip-and-ring voltage to the attached equipment when the call...
Administration Guide
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ATA Voice Field Reference Regional page B Cfwd All Deact Code Cfwd Busy Act Code Cancels call forwarding of no -answer calls to the extension specified after the activation code. The default is *90. Blocks the last inbound call . Accepts the last outbound call . Disables...last outbound call . CW Deact Code The default is *57. Cancels call forwarding of all calls are enabled. The default is *56. Forwards no -answer calls. The default is *91. Cancels call waiting on all calls. Cfwd Last Act Code Forwards the last inbound or outbound calls to the extension...
ATA Voice Field Reference Regional page B Cfwd All Deact Code Cfwd Busy Act Code Cancels call forwarding of no -answer calls to the extension specified after the activation code. The default is *90. Blocks the last inbound call . Accepts the last outbound call . Disables...last outbound call . CW Deact Code The default is *57. Cancels call forwarding of all calls are enabled. The default is *56. Forwards no -answer calls. The default is *91. Cancels call waiting on all calls. Cfwd Last Act Code Forwards the last inbound or outbound calls to the extension...
Administration Guide
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...). The default is no . If the caller does not respond to the refresh message, the ATA device ends this value is not zero, it auto-answers incoming calls and streams audio RTP packets to 255 seconds (0 means that the session refresh is still active. SAS Enable SAS DLG Refresh Intvl To...
...). The default is no . If the caller does not respond to the refresh message, the ATA device ends this value is not zero, it auto-answers incoming calls and streams audio RTP packets to 255 seconds (0 means that the session refresh is still active. SAS Enable SAS DLG Refresh Intvl To...
Administration Guide
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...to perform an attended transfer operation by referring the other call leg. To use this feature, select yes. This field is found on the SPA2102 only. Default is disabled, the ATA device performs an attended transfer operation by ending the current call leg and performing a blind transfer of mixing...on the Phone pages only. To use this feature, select yes. The IP address of the local address enclosed in the FROM of the auto-answering streaming audio server. Voice tab > Line page > Call Feature Settings section On the SPA8800, these settings are configured on -hold is disabled if...
...to perform an attended transfer operation by referring the other call leg. To use this feature, select yes. This field is found on the SPA2102 only. Default is disabled, the ATA device performs an attended transfer operation by ending the current call leg and performing a blind transfer of mixing...on the Phone pages only. To use this feature, select yes. The IP address of the local address enclosed in the FROM of the auto-answering streaming audio server. Voice tab > Line page > Call Feature Settings section On the SPA8800, these settings are configured on -hold is disabled if...