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Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide (SIP) (2.15, Rev B0)
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Index
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Using SIP Supplementary Services
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Preface
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Cisco Analog Telephone Adapter Overview
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Installing the Cisco ATA
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Configuring the Cisco ATA for SIP
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Advanced SIP Services
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Parameters and Defaults
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Call Commands
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Configuring and Debugging Fax Services
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Upgrading the Cisco ATA Signaling Image
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Troubleshooting
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Voice Menu Codes
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Cisco ATA Specifications
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SIP Call Flows
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Glossary
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FeedbackTable Of Contents
Basic and Additional SIP Services
Comfort Noise During Silence Period When Using G.711
Configurable Hook Flash Timing
Configurable Mixing of Call Waiting Tone and Audio
Disabling Access To The Web Interface
Stuttering Dial Tone on Unconditional Call Forward
User Configurable Call Waiting Permanent Default Setting
User Configurable Timeout On No Answer for Call Forwarding
Setting Up and Placing a Call Without Using a SIP Proxy
Complete Reference Table of all Cisco ATA SIP Services
Basic and Additional SIP Services
This section provides information about key basic and additional SIP services that the Cisco ATA supports:
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Important Basic SIP Services—This section includes a list of parameters that you must configure in order for the Cisco ATA to function in a SIP environment.
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Important Basic SIP Services
This section provides descriptions and cross references for configuring required SIP parameters and also for configuring other important basic SIP services:
Required Parameters
If the UseSIP parameter is set to 1 (you are using the SIP protocol), you need to supply values for the required SIP parameters shown in Table 4-1. The Parameter column provides the name of the parameter and a cross reference which provides a more-detailed description of the parameter.
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Establishing Authentication
The Cisco ATA supports two levels of authentication, depending on the setting of the UseLoginID parameter:
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Related Configuration Parameters
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Setting the Codec
The LBRCodec (low-bit-rate codec) parameter determines whether the G.723 or G.729A codec, in addition to G.711A-law and G.711ยต-law, can be used for receiving and transmitting. For configuration information, see the "LBRCodec" section.
Configuring Refresh Interval
When the value specified in the CfgInterval parameter is reached, the Cisco ATA attempts to refresh its configuration file from the TFTP server. By opening a web page for the Cisco ATA, you can perform a refresh before the scheduled refresh. Set the CfgInterval parameter to an interval value (in seconds) for refreshing the Cisco ATA configuration file. Cisco recommends that the interval be semi-random to prevent many simultaneous contacts with the TFTP server. For more information, see the "CfgInterval" section.
When the Cisco ATA contacts the TFTP server, it also checks to see if an upgrade signaling image has been placed on the TFTP server. If such an image exists, the Cisco ATA will download this image.
Additional SIP Services
This section describes additional SIP services and, where applicable, provides configuration information and cross references to the parameters for configuring these services. These services are listed alphabetically.
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Advanced Audio Configuration
The UDPTOS (specifies the default IP precedence of UDP packets) and AudioMode (audio operating mode) parameters allow you to tune audio configuration.
Related Parameters
Billable Features
You can customize specific features on a subscription basis by changing the values of specific bits in several different parameters. Table 4-2 contains a list of billable features and their related parameters:
Table 4-2 Billable Features and Related Parameters
Call Conferencing
Call Forwarding
Call Transfer
Call Waiting
Caller ID
Call Return
Polarity
Voice Mail Indicator
Note
Comfort Noise During Silence Period When Using G.711
When silence suppression is turned on in ITU G.711, the Cisco ATA calculates and transmits its noise level to the far end to enable the remote endpoint to generate the appropriate amount of comfort noise. This provides the remote user with a similar experience to that of a PSTN call and prevents silent gaps when neither party is talking.
Related Parameter
AudioMode—Bit 0 disables/enables silence suppression.
Configurable Hook Flash Timing
This feature provides the ability to adjust the hook-flash timing to meet local requirements.
Related Parameter
SigTimer—Bits 26 and 27 are for configuring the minimum on-hook time required for a hook flash event, and bits 28 through 31 are for configuring maximum on-hook time.
Configurable Mixing of Call Waiting Tone and Audio
This feature allows the call-waiting tone to be mixed with the audio in an active call. Therefore, the call-waiting tone will sound without a pause in the audio.
Related Parameter
ConnectMode—Bit 24
Configurable On-hook delay
This feature is available only for the recipient (callee) of a call. If the callee picks up the phone and then later hangs up to retrieve another call, the hang-up is not considered on-hook until the specified delay expires.
Related Parameter
FeatureTimer—Bits 8 to 12
Debugging Diagnostics
You can use the following parameters to troubleshoot operation issues:
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Dial Plan
You can set specific dial plan rules and timeout values. Many of these values are determined on a country-by-country basis.
Related Parameter
Disabling Access To The Web Interface
To prevent tampering and unauthorized access to the Cisco ATA configuration, the Cisco ATA built-in web server can be disabled.
Related Parameter
OpFlags—Bit 7
Distinctive Ringing
This feature allows a user to identify a caller based on the ringing pattern the user selects for the incoming number.
This feature is dependent on the proxy or remote UA, including the Alert-Info header with the appropriate value in the INVITE message. The Cisco ATA supports standard distinctive ringing pattern 1 to 5 as defined in the standard GR-506-CORE.
The following Alert-Info header values are allowed:
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If the Alert-Info header value is not recognized, the Cisco ATA plays the regular ring tone, Bellcore-dr1.
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DNS SRV Support
The Cisco ATA supports DNS SRV lookup for the SIP proxy server. If the GkOrProxy parameter value begins with _sip._udp. or sip.udp., the Cisco ATA performs a DNS SRV lookup for the SIP proxy server. A DNS SRV lookup results in one of the following conditions:
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Related Parameters
Hardware Information Display
Cisco ATA hardware information is displayed in the lower-left corner of the Cisco ATA Web configuration page.
NAT Gateway
Network Address Translation (NAT) supports port mapping and forwarding to standard default SIP signaling port 5060 and media base port 16384, or other ports as configured in the Cisco ATA. Media ports are evenly numbered from the base port. NAT must support multiple port mappings. The Cisco ATA can use up to four media ports to handle conference calls on both lines. For example, if media base port 16384 is used for one call, the next call uses port 16386 and other calls will use ports 16388 and 16390.
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To configure the Cisco ATA to work in a NAT environment, modify the following parameters:
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NAT/PAT Translation
To maintain Network Address Translation/Port Address Translation (NAT/PAT) for a session, the Cisco ATA can be configured to periodically send a dummy UDP packet to a server (the Cisco ATA does not expect any response from the server).
Related Parameters
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Network Timing
You can fine tune your network timing with the following parameters:
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OutBoundProxy Support
If the OutBoundProxy parameter is a fully qualified domain name (FQDN), and DNS returns multiple IP addresses, the first IP address is used as the primary outbound proxy and the second IP address as the secondary outbound proxy. If OutBoundProxy is an IP address or if DNS returns only one IP address, then a backup outbound proxy is not available. The AltGkTimeOut parameter determines the backup proxy timeout value for the outbound proxy.
If the backup proxy fails, the Cisco ATA automatically switches back to the primary proxy if the unit has been using the backup proxy for at least 30 seconds. This effectively prevents the Cisco ATA from switching indefinitely between failing primary and failing backup proxies for the same transactions.
Switching between primary and secondary proxies can occur only for initial INVITE and REGISTER requests. Other requests, such as CANCEL, BYE, ACK, and re-INVITE, do not retry the backup proxy but give up if the current proxy fails.
When OutBoundProxy is enabled, the Cisco ATA determines whether to retry to connect with the backup OutBoundProxy or backup SIP proxy if the INVITE or REGISTER requests fail. If the reason for failure is an ICMP error (such as an unreachable host), the Cisco ATA retries with the backup outbound proxy. If failure is due to timeout while waiting for a response or a 5xx response, the Cisco ATA retries the backup SIP proxy.
Related Parameter
Progress Tones
Values for the following parameters (all defined in the "Call-Progress Tone Parameters" section) must be determined based on the country in which the Cisco ATA is located:
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Receiver-tagged VIA header
You can disable or enable the processing the received = parameter in the Via header. This feature is disabled by default.
Related Parameter
ConnectMode—Bit 22
Repeat Dialing on Busy Signal
This feature allows the Cisco ATA to repeatedly call a busy number at a periodic interval for a specific length of time. Both the interval and total time can be specified by the user.
To use this feature, configure FeatureTimer bits 0-7 and add the new command/action values "#37#;kA" to the existing "H" context and "5;jA" to the existing "S" context in the CallCmd parameter.
This feature is invoked by pressing 5 after the busy tone sounds. The caller then gets a beep confirmation followed by silence. When the subscriber hangs up, the Cisco ATA starts to redial at the interval specified in FeatureTimer bits 4-7. When the called party rings, the caller is notified with a special ring. If the called party picks up the call first, the called party receives a ringback. If the caller picks up the call first, the caller receives the ringback. This feature is automatically cancelled when the called party rings.
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Related Parameters
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Parameter: CallCmdContext: S (may also include 'a' or 'b')Command/action: 5;jADescription: This context command adds the service activation code to enable repeat dialing.Parameter: CallCmdContext: HCommand/action: #37#;kADescription: This context command adds the service deactivation code to disable repeat dialing
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SIP Proxy Server Redundancy
SIP proxy server redundancy can be enabled by entering a fully qualified domain name (FQDN) or IP address (and optional port number) in the GkOrProxy and AltGk parameters, and by configuring the AltGkTimeOut parameter. If you provide hostnames for GkOrProxy or AltGk, the names are resolved by the configured DNS. DNS results are hard-coded in cache memory for 10 minutes.
If DNS returns multiple IP addresses, the Cisco ATA uses only the first IP address. If AltGk is set to 0 (disabled) and DNS returns two or more IP addresses for GkOrProxy, then the Cisco ATA uses the first IP address as the primary proxy and the second IP address as the secondary proxy. If GkOrProxy is an IP address or DNS returns one IP address, then the backup SIP proxy is not available. A special case exists if GkOrProxy and AltGk are the same values and are not IP addresses. In this case, the AltGk parameter is assumed to have the value 0.
Related parameters
Stuttering Dial Tone on Unconditional Call Forward
If unconditional call forwarding is enabled, the Cisco ATA plays a continuous stuttering dial tone when the telephone handset is picked up. This reminds the user that all incoming calls are forwarded to another number. For more information, see the "Call Forwarding in the United States" section and the "Call Forwarding in Sweden" section.
User Configurable Call Waiting Permanent Default Setting
This feature allows you to specify the default call-waiting setting for every call on a permanent basis by means of the service activation and deactivation codes.
Related Parameter
ConnectMode—Bit 23
User Configurable Timeout On No Answer for Call Forwarding
This feature allows you to specify the timeout before a call is forwarded to another number on no answer.
This feature is activated by entering the service activation code followed by the phone number and delay. The entry sequence is as follows:
<Service Activation Code> <Phone Number> * <Delay> #Delay can be from 1 to 255 seconds. If the delay is zero (0) or not provided by the user, the delay specified in the SigTimer parameter (bits 20-25), which has a default value of 20 seconds, is in effect.
Example
Using the U.S. Call Command parameter string, the U.S. service activation code is #75 and the deactivation code is #73.
To forward calls to the number 555-1212 after a no-answer for 15 seconds, enter the following:
#755551212*15#To deactivate this feature, enter the following:
#73Related Parameter
SigTimer—Bits 20 to 25
Setting Up and Placing a Call Without Using a SIP Proxy
The Cisco ATA supports direct IP-to-IP calls without using a SIP proxy. When a call is placed, the Cisco ATA sends the INVITE request directly to the remote user agent and exepcts the usual 100/180/200 responses from the user agent.
This section contains the following topics:
Configuration
To perform the necessary configuration of the Cisco ATA, follow this procedure:
Procedure
Step 1
Step 2
where Cisco_ATA_IP_address is the IP address of your Cisco ATA. This takes you to the Cisco ATA Web configuration page.
Step 3
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Step 4
Placing an IP Call
To place an IP call, dial the telephone number and the IP address of the remote user agent. The dial format is shown below:
Dial Format
<phone number>**<ipaddress>#
Use the star (*) key on the telephone keypad to represent the dot (.) in an IP address. Use the pound (#) key on the telephone keypad to terminate the dial string and place the call.
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Example
To place a call to a user agent with an ID of 408-555-1212 at IP address 192.168.1.100, you would enter the following string on your telephone keypad:
4085551212**192*168*1*100#
Complete Reference Table of all Cisco ATA SIP Services
Table 4-3 is a reference table that lists all configurable features for the Cisco ATA (using SIP), and includes links to the detailed descriptions of the parameters used for configuring these features.
