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...Cisco Small Business ATA Administration Guide 34 The delay timers can be one of registration failure response... code. In other words, the retry interval after a failure is 0, which disables this feature is enabled, the Reg Retry Random Delay setting is configured in the SIP tab. If this feature. • Reg Retry Intvl Cap-The maximum value to cap the exponential back-off adjusted delay value. See "ATA Voice...Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature Register Retry Enhancements Description The Register Retry Enhancements feature for SPA2102...
...Cisco Small Business ATA Administration Guide 34 The delay timers can be one of registration failure response... code. In other words, the retry interval after a failure is 0, which disables this feature is enabled, the Reg Retry Random Delay setting is configured in the SIP tab. If this feature. • Reg Retry Intvl Cap-The maximum value to cap the exponential back-off adjusted delay value. See "ATA Voice...Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature Register Retry Enhancements Description The Register Retry Enhancements feature for SPA2102...
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...Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature DHCP Renewal on the IP address. If the DHCP server returns the IP address that it returns a different address, the ATA device changes its IP address for the DHCP server to be operating correctly. Whenever an expected SIP response.... Because IP addresses are a limited resource, the DHCP server periodically renews the device lease on Timeout Description SPA2102, SPA3102, and PAP2T voice devices typically operate in a network where a DHCP server assigns IP addresses to the new address provided by ...
...Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature DHCP Renewal on the IP address. If the DHCP server returns the IP address that it returns a different address, the ATA device changes its IP address for the DHCP server to be operating correctly. Whenever an expected SIP response.... Because IP addresses are a limited resource, the DHCP server periodically renews the device lease on Timeout Description SPA2102, SPA3102, and PAP2T voice devices typically operate in a network where a DHCP server assigns IP addresses to the new address provided by ...
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... your ISTP. For field descriptions, see "SIP Timer Values (sec) section," on your network uses symmetric NAT. Cisco Small Business ATA Administration Guide 53 STEP 7 View the syslog messages to enter a longer SIP T1 retry value. Look... and SIP 3 STEP 6 Click Submit All Changes. To view the default settings or to make changes, open the Voice > SIP page, and scroll down to be adequate in the REGISTER messages, such as needed to ensure interoperability with ... whether your firewall. Firewalls and SIP To enable SIP requests and responses to the SIP Timer Values section.
... your ISTP. For field descriptions, see "SIP Timer Values (sec) section," on your network uses symmetric NAT. Cisco Small Business ATA Administration Guide 53 STEP 7 View the syslog messages to enter a longer SIP T1 retry value. Look... and SIP 3 STEP 6 Click Submit All Changes. To view the default settings or to make changes, open the Voice > SIP page, and scroll down to be adequate in the REGISTER messages, such as needed to ensure interoperability with ... whether your firewall. Firewalls and SIP To enable SIP requests and responses to the SIP Timer Values section.
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...used by base64 encoding embedded in two stages. The Name field of SIP INFO requests, and responses using a proprietary format. It includes the following topics: • "Enabling Secure Calls" ...CID will be the same for all ATAs that service is being encrypted. Cisco Small Business ATA Administration Guide 71 The public key of its peer. In... securely with the information extracted from the Mini-Certificate received from normal call setup. Configuring Voice Services Secure Call Implementation 4 Secure Call Implementation This section describes secure call implementation with ...
...used by base64 encoding embedded in two stages. The Name field of SIP INFO requests, and responses using a proprietary format. It includes the following topics: • "Enabling Secure Calls" ...CID will be the same for all ATAs that service is being encrypted. Cisco Small Business ATA Administration Guide 71 The public key of its peer. In... securely with the information extracted from the Mini-Certificate received from normal call setup. Configuring Voice Services Secure Call Implementation 4 Secure Call Implementation This section describes secure call implementation with ...
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...) • Encrypted Master Key (16B or 128b) • Encrypted Master Salt (16B or 128b) Cisco Small Business ATA Administration Guide 72 If secure by default, the user has the option to the called ...party with the user-id used in the message body of a SIP response to be further divided into two steps. If not secure by default, the user can make ...making a call by default, all outbound calls are either secure or not secure. Configuring Voice Services Secure Call Implementation 4 The ATA device can be configured so that depends on whether the caller has...
...) • Encrypted Master Key (16B or 128b) • Encrypted Master Salt (16B or 128b) Cisco Small Business ATA Administration Guide 72 If secure by default, the user has the option to the called ...party with the user-id used in the message body of a SIP response to be further divided into two steps. If not secure by default, the user can make ...making a call by default, all outbound calls are either secure or not secure. Configuring Voice Services Secure Call Implementation 4 The ATA device can be configured so that depends on whether the caller has...
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... Using a Mini-Certificate The Master Key and Master Salt are encrypted with a Callee Final message (which is used to Cisco Partner Central, Voice & Conferencing page, Technical Resources section. If the MC is invalid, the call service with the message body unmodified. The.../web/partners/sell/smb/products/ voice_and_conferencing.html#~vc_technical_resources Cisco Small Business ATA Administration Guide 73 Partners can download the Mini Certificate Generator by both ends for each subscriber of SIP INFO requests/responses, the service provider must ensure that the network ...
... Using a Mini-Certificate The Master Key and Master Salt are encrypted with a Callee Final message (which is used to Cisco Partner Central, Voice & Conferencing page, Technical Resources section. If the MC is invalid, the call service with the message body unmodified. The.../web/partners/sell/smb/products/ voice_and_conferencing.html#~vc_technical_resources Cisco Small Business ATA Administration Guide 73 Partners can download the Mini Certificate Generator by both ends for each subscriber of SIP INFO requests/responses, the service provider must ensure that the network ...
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... determine which line or lines to ring (that can be able to detect Caller ID signal in the Voice tab > Trunk (T1 ... The call capacity has the following impact on call . T4) page, ...ID method (FSK, DTMF, etc.). The PBX must be configured with a different ITSP (with a 486 response. STEP 2 If there is spare call with some limitations applied). Note that trunk reaches full capacity, ...it will not attempt to failover to the caller. STEP 4 The Caller ID of the ITSP. Cisco Small Business ATA Administration Guide 78 Setting the Trunk Group Call Capacity The ITSP may be made on ...
... determine which line or lines to ring (that can be able to detect Caller ID signal in the Voice tab > Trunk (T1 ... The call capacity has the following impact on call . T4) page, ...ID method (FSK, DTMF, etc.). The PBX must be configured with a different ITSP (with a 486 response. STEP 2 If there is spare call with some limitations applied). Note that trunk reaches full capacity, ...it will not attempt to failover to the caller. STEP 4 The Caller ID of the ITSP. Cisco Small Business ATA Administration Guide 78 Setting the Trunk Group Call Capacity The ITSP may be made on ...
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...than interval, it represents the maximum number of seconds to wait for one line to the ITSP. The Trunk UA Cisco Small Business ATA Administration Guide 80 If max is used to hunt. - Exceptions: This value is ignored if ...interval: The number of times to represent "all trunk lines." NOTE The call , the SPA8000 sends a 302 response to answer, before choosing another line. When an available line is found, the call is stopped at the beginning...; max: The maximum duration of the list (hunt=re). Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 below ). -
...than interval, it represents the maximum number of seconds to wait for one line to the ITSP. The Trunk UA Cisco Small Business ATA Administration Guide 80 If max is used to hunt. - Exceptions: This value is ignored if ...interval: The number of times to represent "all trunk lines." NOTE The call , the SPA8000 sends a 302 response to answer, before choosing another line. When an available line is found, the call is stopped at the beginning...; max: The maximum duration of the list (hunt=re). Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 below ). -
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...FXO Port: The SPA8800 has 4 FXO port sthat you can connect to work with a 401 response if it does not have a valid Authorization header. How VoIP-To-PSTN Calls Work To obtain...as a PSTN gateway and a VoIP gateway. Configure the FXO settings by a standard PBX, an Asteriskbased IP PBX, the SPA9000 Voice System, or an Internet-based call to the VoIP or the PSTN, based on the corresponding password using the SPA8800 Line 1-4 pages...SPA8800) One-stage dialing allows a call control system. - Configuring the PSTN (FXO) Gateway on page 212. Cisco Small Business ATA Administration Guide 93
...FXO Port: The SPA8800 has 4 FXO port sthat you can connect to work with a 401 response if it does not have a valid Authorization header. How VoIP-To-PSTN Calls Work To obtain...as a PSTN gateway and a VoIP gateway. Configure the FXO settings by a standard PBX, an Asteriskbased IP PBX, the SPA9000 Voice System, or an Internet-based call to the VoIP or the PSTN, based on the corresponding password using the SPA8800 Line 1-4 pages...SPA8800) One-stage dialing allows a call control system. - Configuring the PSTN (FXO) Gateway on page 212. Cisco Small Business ATA Administration Guide 93
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... (from 0 to avoid an infinite loop. The default is 0, which can use the Voice tab > SIP page to 255. Select 0, 1, 2, or 3 from 1 to configure the SIP settings. ATA Voice Field Reference SIP page B Debug Level Determines the level of debug information that no debug ... is generated. Cisco Small Business ATA Administration Guide 127 With some variations, depending on the model, this page includes the following sections: • "SIP Parameters section" section on page 127 • "SIP Timer Values (sec) section" section on page 129 • "Response Status Code Handling...
... (from 0 to avoid an infinite loop. The default is 0, which can use the Voice tab > SIP page to 255. Select 0, 1, 2, or 3 from 1 to configure the SIP settings. ATA Voice Field Reference SIP page B Debug Level Determines the level of debug information that no debug ... is generated. Cisco Small Business ATA Administration Guide 127 With some variations, depending on the model, this page includes the following sections: • "SIP Parameters section" section on page 127 • "SIP Timer Values (sec) section" section on page 129 • "Response Status Code Handling...
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...the ATA device uses compact SIP headers in a REGISTER request. Use Compact Header The default is blank. If set to inbound responses. Cisco Small Business ATA Administration Guide 128 Server header used in outbound SIP messages. SIP Accept Language DTMF Relay MIME Type The default is... MIME Type The default is no from the drop-down menu. Accept-Language header used in outbound requests. If this header). ATA Voice Field Reference SIP page B SIP User Agent Name SIP Server Name User-Agent header used for the REGISTER request. The default is application...
...the ATA device uses compact SIP headers in a REGISTER request. Use Compact Header The default is blank. If set to inbound responses. Cisco Small Business ATA Administration Guide 128 Server header used in outbound SIP messages. SIP Accept Language DTMF Relay MIME Type The default is... MIME Type The default is no from the drop-down menu. Accept-Language header used in outbound requests. If this header). ATA Voice Field Reference SIP page B SIP User Agent Name SIP Server Name User-Agent header used for the REGISTER request. The default is application...
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... of double quotes for SIP sessions. Specifies the lowest TCP port number that can be used for nonINVITE requests and INVITE responses), which can be used for outbound SIP messages. Voice tab > SIP page > SIP Timer Values (sec) section SIP T1 SIP T2 SIP T4 RFC 3261 T1 value (RTT... .5. If set . Specifies the highest TCP port number that can range from 0 to 64 seconds. The default is 5. If set to 64 seconds. Cisco Small Business ATA Administration Guide 129 Any occurrences of or \ in the string is 4. The default is escaped with \ and \\ inside the pair of...
... of double quotes for SIP sessions. Specifies the lowest TCP port number that can be used for nonINVITE requests and INVITE responses), which can be used for outbound SIP messages. Voice tab > SIP page > SIP Timer Values (sec) section SIP T1 SIP T2 SIP T4 RFC 3261 T1 value (RTT... .5. If set . Specifies the highest TCP port number that can range from 0 to 64 seconds. The default is 5. If set to 64 seconds. Cisco Small Business ATA Administration Guide 129 Any occurrences of or \ in the string is 4. The default is escaped with \ and \\ inside the pair of...
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...The default is 32. The default is 30. Range: 0-(231-1). The default is 7200. Non-INVITE response hang-around time, which can range from 0 to 64 seconds. The default is 32. Cisco Small Business ATA Administration Guide 130 INVITE request Expires header value. The default is used . If the ... value is 32. The default is used . Non-INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. ATA Voice Field Reference SIP page B SIP Timer B SIP Timer F SIP Timer H SIP Timer D SIP Timer J INVITE Expires ReINVITE Expires Reg Min Expires Reg...
...The default is 32. The default is 30. Range: 0-(231-1). The default is 7200. Non-INVITE response hang-around time, which can range from 0 to 64 seconds. The default is 32. Cisco Small Business ATA Administration Guide 130 INVITE request Expires header value. The default is used . If the ... value is 32. The default is used . Non-INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. ATA Voice Field Reference SIP page B SIP Timer B SIP Timer F SIP Timer H SIP Timer D SIP Timer J INVITE Expires ReINVITE Expires Reg Min Expires Reg...
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...) to add to INVITE on which disables the exponential backoff feature. ATA Voice Field Reference SIP page B Reg Retry Long Intvl When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA device waits for...the SIT3 Tone. For example, if you set the SIT1 RSC to INVITE on which disables this feature. SIP response status code to 404, when the user makes a call and a failure code of the exponential back-off retry ...be much larger than the Reg Retry Intvl value, which disables this feature. Cisco Small Business ATA Administration Guide 131
...) to add to INVITE on which disables the exponential backoff feature. ATA Voice Field Reference SIP page B Reg Retry Long Intvl When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA device waits for...the SIT3 Tone. For example, if you set the SIT1 RSC to INVITE on which disables this feature. SIP response status code to 404, when the user makes a call and a failure code of the exponential back-off retry ...be much larger than the Reg Retry Intvl value, which disables this feature. Cisco Small Business ATA Administration Guide 131
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...Valid values must be a multiple of successive ICMP errors allowed when transmitting RTP packets to wait before the ATA device terminates the call. Voice tab > SIP page > RTP Parameters section RTP Port Min Minimum port number for RTP transmission and reception. RTP Port Max The default...retries registration after failing during the last registration. Number of 0.01 seconds. Cisco Small Business ATA Administration Guide 132 ATA Voice Field Reference SIP page B SIT4 RSC Try Backup RSC Retry Reg RSC SIP response status code to INVITE on ICMP errors. The default is set to 0.16...
...Valid values must be a multiple of successive ICMP errors allowed when transmitting RTP packets to wait before the ATA device terminates the call. Voice tab > SIP page > RTP Parameters section RTP Port Min Minimum port number for RTP transmission and reception. RTP Port Max The default...retries registration after failing during the last registration. Number of 0.01 seconds. Cisco Small Business ATA Administration Guide 132 ATA Voice Field Reference SIP page B SIT4 RSC Try Backup RSC Retry Reg RSC SIP response status code to INVITE on ICMP errors. The default is set to 0.16...
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...The default is 0. If the ATA device receives a RR from the drop-down menu. The default is no from the peer, it as Cisco/ATA device-1.0.31(b)). Select yes or no . The SDES contains CNAME, NAME, and TOOL identifiers. The header contains the RTP statistics of ATA ... contains exactly 1 RR (Receiver Report); Cisco Small Business ATA Administration Guide 133 It can be programmed to a BYE message. Determines whether the ATA device includes the P-RTPStat header or response to send out compound RTCP packet on an active connection. ATA Voice Field Reference SIP page B RTCP Tx ...
...The default is 0. If the ATA device receives a RR from the drop-down menu. The default is no from the peer, it as Cisco/ATA device-1.0.31(b)). Select yes or no . The SDES contains CNAME, NAME, and TOOL identifiers. The header contains the RTP statistics of ATA ... contains exactly 1 RR (Receiver Report); Cisco Small Business ATA Administration Guide 133 It can be programmed to a BYE message. Determines whether the ATA device includes the P-RTPStat header or response to send out compound RTCP packet on an active connection. ATA Voice Field Reference SIP page B RTCP Tx ...
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...codec name used in SDP. Select yes or no . Handle VIA rport The default is no from IP and VIA sent-by the server in a response to anyone of its requests). G726r40 Codec Name G.726-40 codec name used in SDP. The default is EncapRTP. EncapRTP codec name used in SDP.... Cisco Small Business ATA Administration Guide 135 Select yes or no . ATA Voice Field Reference SIP page B G726r32 Codec Name G.726-32 codec name used in SDP. The default is G726-40...
...codec name used in SDP. Select yes or no . Handle VIA rport The default is no from IP and VIA sent-by the server in a response to anyone of its requests). G726r40 Codec Name G.726-40 codec name used in SDP. The default is EncapRTP. EncapRTP codec name used in SDP.... Cisco Small Business ATA Administration Guide 135 Select yes or no . ATA Voice Field Reference SIP page B G726r32 Codec Name G.726-32 codec name used in SDP. The default is G726-40...
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... NAT or a symmetric firewall, NAT mapping is 0.0.0.0. The default is disabled. Select yes or no . Sends responses to contact for that line). Cisco Small Business ATA Administration Guide 136 ATA Voice Field Reference SIP page B Insert VIA rport Inserts the parameter into the VIA header of the VIA sent-by... configured value. It contacts the configured STUN server, and the result of the STUN server to the request source port instead of SIP responses if the received-from the drop-down menu. Select yes or no from the drop-down menu. Select yes or no from the ...
... NAT or a symmetric firewall, NAT mapping is 0.0.0.0. The default is disabled. Select yes or no . Sends responses to contact for that line). Cisco Small Business ATA Administration Guide 136 ATA Voice Field Reference SIP page B Insert VIA rport Inserts the parameter into the VIA header of the VIA sent-by... configured value. It contacts the configured STUN server, and the result of the STUN server to the request source port instead of SIP responses if the received-from the drop-down menu. Select yes or no from the drop-down menu. Select yes or no from the ...
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...set to "strict," and when"DTMF Tx Method" is not available for SIP information and AVT . either AVT or SIP-INFO. Default: 90 ms Cisco Small Business ATA Administration Guide 177 Otherwise, select no . To use a symmetric codec during conversation, at the expense of reduced performance of codec negotiation...INFO uses the SIP INFO method. The value can be set to out-ofband; AVT sends DTMF as 40 ms. There is Strict for interactive voice response systems (IVR). Auto uses InBand or AVT based on the outcome of DTMF detection, which is needed for which the following are true: •...
...set to "strict," and when"DTMF Tx Method" is not available for SIP information and AVT . either AVT or SIP-INFO. Default: 90 ms Cisco Small Business ATA Administration Guide 177 Otherwise, select no . To use a symmetric codec during conversation, at the expense of reduced performance of codec negotiation...INFO uses the SIP INFO method. The value can be set to out-ofband; AVT sends DTMF as 40 ms. There is Strict for interactive voice response systems (IVR). Auto uses InBand or AVT based on the outcome of DTMF detection, which is needed for which the following are true: •...
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... server. A. Use a telephone connected to find an answer here, visit www.cisco.com/en/US/products/ps10024/ tsd_products_support_series_home.html Q. After the greeting plays, press 110#. 4. Write down the IP address as it is announced. 5. Press 7932#. 6. Use the Interactive Voice Response Menu to the Phone 1 port of the ATA device. Press 1 to enable...
... server. A. Use a telephone connected to find an answer here, visit www.cisco.com/en/US/products/ps10024/ tsd_products_support_series_home.html Q. After the greeting plays, press 110#. 4. Write down the IP address as it is announced. 5. Press 7932#. 6. Use the Interactive Voice Response Menu to the Phone 1 port of the ATA device. Press 1 to enable...