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Table Of Contents
Basic and Additional SIP Services
Call Forwarding Setting Removal Using HTTP
Comfort Noise During Silence Period When Using G.711
Configurable Hook Flash Timing
Configurable Mixing of Call Waiting Tone and Audio
Configurable Reboot of Cisco ATA
Disabling Access To The Web Interface
Display-Name Support for Caller ID
Obtaining Network Status Before and After Getting IP Connectivity
Real-Time Transport Protocol (RTP) Statistics Reporting
Redundant Proxy Support for BYE/CANCEL Request
Retransmitting SIP requests and SIP Responses
Setting Up and Placing a Call Without Using a SIP Proxy
Status of Phone Service Using HTTP
Stuttering Dial Tone on Unconditional Call Forward
Toll Restrictions for Call Forwarding and Outgoing Calls
User Configurable Call Waiting Permanent Default Setting
User Configurable Timeout On No Answer for Call Forwarding
Voice Prompt Confirmation for Call Waiting and Call Forwarding
XML Pages of Cisco ATA Information
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:
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Required Parameters
If 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.
Note
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
Setting the Codec
The LBRCodec (low-bit-rate codec) parameter determines whether the G.723, G.726 or G.729A codec, in addition to G.711A-law and G.711µ-law, can be used for receiving and transmitting. For configuration information, see the "LBRCodec" section on page 5-32.
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 on page 5-6.
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 TOS (specifies the precedence and delay of audio and signaling IP 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
PaidFeatures, page 5-36, CallFeatures, page 5-35, ConnectMode, page 5-41, SigTimer, page 5-40
Call Transfer
Call Waiting
PaidFeatures, page 5-36, CallFeatures, page 5-35, SigTimer, page 5-40
Caller ID
PaidFeatures, page 5-36, CallFeatures, page 5-35, CallerIdMethod, page 5-49
Call Return
ConnectMode, page 5-41, PaidFeatures, page 5-36, CallFeatures, page 5-35
Polarity
Voice Mail Indicator
Note
Call Forwarding Setting Removal Using HTTP
The service provider can remotely reset a call forwarding setting for which a subscriber configured an incorrect phone number to receive fowarded calls.
The service provider issues the following command, which removes call forwarding settings for both Cisco ATA phone lines:
http://ipaddress/resetcfwd/where ipaddress is the IP address of the Cisco ATA whose call forwarding numbers are being removed.
This Web page is password protected. Once the service provider issues this command, this Web page shows that the current call-waiting and call-forwarding settings are N/A.
Call-Waiting Hang-Up Alert
This feature provides an audible alert (ringtone) whenever the user inadvertently hangs up from a call-waiting call while an active call is still on hold.
This section contains the following topics:
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Enabling the Call-Waiting Hang-Up Alert Feature
To enable the call-waiting hang-up alert feature, perform the following steps:
Procedure
Step 1
Step 2
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Default Behavior of Call-Waiting Calls
Without the call-waiting hang-up alert feature enabled, both the call-waiting call and active call are disconnected as soon as the user hangs up the phone.
To check whether this default behavior is in effect, search for the appearance of the string h;HH within the sequence of call-waiting call commands. The context-identifier K denotes the beginning of the call-waiting call commands.
With this default behavior, the call-waiting call command string could be one of the following examples:
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For more information about call commands, see "Call Commands"
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, page 5-32—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, page 5-40—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, page 5-41—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, page 5-38—Bits 8 to 12
Configurable Reboot of Cisco ATA
The Cisco ATA continuously monitors its Ethernet connection to the switch or hub. If this connection is broken, the Cisco ATA starts an internal timer that runs until a configurable timeout period expires. Once the timeout value is reached, the Cisco ATA automatically reboots. This timeout value is configured by using the FeatureTimer2 configuration parameter. For more information, see the "FeatureTimer2" section on page 5-39.
Diagnostics for Debugging
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 Parameters
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, page 5-45—Bit 7
Display-Name Support for Caller ID
For caller ID purposes, you can configure a name to correspond to the phone number of the Cisco ATA input ports. This name will be displayed at the remote endpoint when a call originates from this Cisco ATA.
Related Parameters
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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.
Note
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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Obtaining Network Status Before and After Getting IP Connectivity
Using voice configuration menu code 3123#, you can obtain basic network status to use for diagnostic purposes prior to getting IP connectivity. For detailed information, see the "Obtaining Network Status Prior to Getting IP Connectivity" section.
Use the Cisco ATA Stats Web page (http://<Cisco ATA IP address>/stats) to display network status information after obtaining IP connectivity. For detailed information, see the "Obtaining Network Status After Getting IP Connectivity" section.
Privacy Options
The privacy options described in this section provide users with stricter control over the appearance of their caller line identification at the SIP message level. These options not only protect the user's anonymity at the caller site but also prevent network-level sniffer-type applications from gaining access to restricted information that is in transit.
This section contains the following topics:
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Network Infrastructure Requirements
For user privacy to work effectively on the Cisco ATA, the proxy or proxies deployed in the IP network must be capable of the following:
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Gateways deployed in the IP Network must be capable of the following:
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Anonymity for Called Party
The Cisco ATA provides an option to use the Anonymous user name in the TO header in all outgoing SIP INVITE requests when the Cisco ATA is the calling party.
To enable this option, set bit 30 of the Cisco ATA ConnectMode parameter to 1. For more information, see the "ConnectMode" section on page 5-41.
This feature guarantees the anonymity of the called party (in the To header) on every call that the Cisco ATA initiates. However, the main intent of this feature is to hide the identity of the called party in the event that the call is diverted to a not-trusted address and caller-ID-restricted is configured on the diverting Cisco ATA. In this case, the proxy has the responsibility of removing user-sensitive data before the call is sent to the not-trusted address.
The Cisco ATA handles the TO header because proxies are not allowed to change the TO header.
Anonymous User Name Support for SIP INVITE Requests
The Cisco ATA provides an option to use the Anonymous user name in the FROM and CONTACT headers and in the o= line (also called the origin line) of the Session Description Protocol (SDP) header in SIP INVITE requests to the far end. The Anonymous user name is used when the following three conditions are met:
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Example Cisco ATA INVITE Messages
The following example SIP INVITE messages show how the Anonymous user name would appear in the context of these messages:
INVITE sip:9401@192.168.2.146:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.2.96:5060;branch=7cca152b-232af34f-3f0b0c31-3cea3a52-1Via: SIP/2.0/UDP 192.168.3.117:5060;received=192.168.3.117Supported: timerFrom: "Anonymous" <sip:Anonymous@192.168.2.96;user=phone>;tag=173234376To: <sip:9401@192.168.2.96;user=phone>Call-ID: 1157628352@192.168.3.117CSeq: 1 INVITEContact: "Anonymous" <sip:Anonymous@192.168.3.117:5060;user=phone;transport=udp>User-Agent: Cisco ATA 186 v3.0.0 atasip (030619A)Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTERExpires: 300Content-Length: 252Content-Type: application/sdpv=0o=Anonymous 6269 6269 IN IP4 192.168.3.117Privacy Token Support for SIP Diversion Header
Proxies typically use the privacy token value contained in the Diversion header of SIP INVITE messages and 3xx Redirection responses to determine whether any of the diverting party's user names should be changed before forwarding the message to untrusted addresses.
This feature applies only when the following two conditions are met:
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Before forwarding the call, the Cisco ATA appends a privacy=[full|off] field to the end of the Diversion header in a 302 Moved Temporarily message.
The value of the privacy=[full|off] field depends on the setting of bit 3 of the CallFeatures and PaidFeatures parameters. (Bit 19 is the applicable bit for the Phone2 port of the Cisco ATA.) This bit is for configuring either the Caller Line Identification Restriction (CLIR) or Caller Line Identification Presentation (CLIP) feature.
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For more information on CLIR and CLIP, see the "CallFeatures" section on page 5-35 and the "PaidFeatures" section on page 5-36.
Progress Tones
Values for the following parameters (all defined in the "Tone Configuration Parameters" section on page 5-53) must be determined based on the country in which the Cisco ATA is located:
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Real-Time Transport Protocol (RTP) Statistics Reporting
To monitor the quality of service for the media stream, you can access RTP packet statistics of the two voice ports and their channels by opening the following page on the Cisco ATA Web server:
<Cisco ATA IP address>/rtps
For detailed information about RTP statistics reporting, see the "Real-Time Transport Protocol (RTP) Statistics Reporting" section.
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, page 5-41—Bit 22
Redundant Proxy Support for BYE/CANCEL Request
The Cisco ATA retries a BYE or CANCEL request using an alternate SIP proxy if the GkOrProxy parameter value is configured with a domain name. The BYE request requires special consideration because the destination can be either the SIP endpoint client or proxy server.
For a SIP user agent client, if a SIP proxy server does not include a Record-Route header in its 200 OK response to an INVITE request, the destination of a BYE request is the SIP URL specified in the Contact header of the response. This URL is usually an IP address; therefore, redundancy is not possible.
For a SIP user agent server, if a SIP proxy server does not include a Record-Route header in the original INVITE request, the destination of a BYE request is the SIP URL specified in the Contact header of the request. This URL is also usually an IP address; therefore, redundancy is not possible.
If a Record-Route header is present in the SIP proxy server 200 OK response to an INVITE request or in an original INVITE request, the BYE request is sent to the first SIP URL specified in the Record-Route header. If the SIP URL is a proxy domain name, then proxy redundancy is possible. Therefore, for SIP proxy redundancy to work for a BYE request, the RequestURL must be a domain name.
Related Parameter
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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Retransmitting SIP requests and SIP Responses
You can configure the number of Cisco ATA transmission attempts for some SIP requests and responses to requests from the SIP user agent. For details on which requests and responses are configurable, see the "MsgRetryLimits" section on page 5-24.
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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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#
SipOutBoundProxy Support
If the SipOutBoundProxy 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 SipOutBoundProxy 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 SipOutBoundProxy is enabled, the Cisco ATA determines whether to retry to connect with the backup SipOutBoundProxy 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
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
SIP Session-Timer Support
The SIP Session Timer is a keepalive mechanism for a SIP session, and is used to determine whether a call is still active when one of the following conditions occurs:
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To avoid a a situation where a user agent waits indefinitely for a BYE message, the user agent sends periodic re-INVITE requests (or session refresh requests) to the Cisco ATA to keep the session alive. The interval betwen these session-refresh requests is negotiated with the use of Session-Expires/Min-SE headers and 422 (Session Interval Too Small) messages. If the Cisco ATA does not receive a session-refresh request before the negotiated interval expires, the session is considered terminated. Both user agents then send BYE messages and disconnect the session.
The Cisco ATA supports session timing only when both the caller and callee support this feature. Also, the Cisco ATA does not support the UPDATE method; therefore, all session-refresh requests are performed by means of the re-INVITE method.
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Related Parameters
The following three parameters are used to configure SIP session timing:
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Status of Phone Service Using HTTP
You can use the following command to provide the status of various call features:
http://ipaddress/service/where ipaddress is the IP address of the Cisco ATA whose status you are checking.
The Web page invoked from this command provides information about the following features:
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For all features except call waiting, the information on the Web page either provides the applicable phone number, which also means that the feature has been activated, or N/A is shown if the feature has not been activated.
STUN Support
The Cisco ATA supports a Simple Traversal of UDP through NAT (STUN) client, as described in RFC 3489. The Cisco ATA obtains the IP address and port mappings of the NAT and uses them accordingly in a SIP message.
This section contains the following topics:
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Types of NATs
Four types of NATs can be used:
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A STUN server can help facilitate traversing through most NATs, except for symmetric NATs.
NAT Traversal
To help facilitate traversal through a NAT, the Cisco ATA uses the same signaling port for transmitting and receiving SIP messages and the same media port for transmitting and receiving media. Before an external host can send a packet to the Cisco ATA behind a NAT, the Cisco ATA must already have sent a packet through the NAT.
An external host can communicate with a Cisco ATA behind a full cone NAT by sending packets to the mapped port. If the Cisco ATA is behind a restricted cone NAT, an external host would have to send packets from the same IP address that it used to receive packets from the Cisco ATA.
If the Cisco ATA is behind a port-restricted cone NAT, an external host would have to send packets from the same IP address and port it used to receive packets from the Cisco ATA. Because a symmetric NAT creates different mapping for every packet with a different destination IP address and/or port, the Cisco ATA cannot traverse through this type of NAT.
If properly configured, the Cisco ATA on power up contacts the specified STUN server to obtain the IP address and port mappings of the NAT before registering with the registration server. The Cisco ATA substitutes this mapping information into the Via and Contact headers. Each time the Cisco ATA sends an INVITE message, the Cisco ATA obtains IP address and port mappings from the STUN server and substitutes this mapping information into the Via, Contact, and SDP c= and m= headers. This allows the SIP proxy server and remote user agents to to communicate with the Cisco ATA through most NATs.
STUN Configuration Parameters
Two parameters control the operation of the Cisco ATA with a STUN server:
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For more information, see the "NatTimer" section on page 5-22.
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For more information, see the "NatServer" section on page 5-22.
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.
Toll Restrictions for Call Forwarding and Outgoing Calls
You can configure the Cisco ATA to block certain numbers from call forwarding and to display certain numbers for caller ID.
Related Dial Plan Rules
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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, page 5-41—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, page 5-40—Bits 20 to 25
Voice Prompt Confirmation for Call Waiting and Call Forwarding
You can configure the Cisco ATA to automatically call a voice announcement server whenever the status of call-waiting or call-forwarding services changes. The telephone number of the server to which the Cisco ATA will send an INVITE request is specified by a configurable base number and several pre-assigned extension numbers. The extension numbers correspond to the service and the state change of the service.
You must configure the following information:
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Base Number
The base number is the first part of a number that the Cisco ATA calls, as specified in the dial plan using rule `B.' To set this base number using rule plan `B', you would use `B' followed by the desired base number. If, for example, the desired base number is 1234, you would add the rule `B1234' to your dial plan.
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The telephone number that the Cisco ATA will call will always consist of the base number followed by a two-digit extension. The extensions corresponding to the service type and service transition, as shown below. These extensions are not configurable. If the administrator has configured a base number of 1234 and call forward on busy is enabled, the called number is 123403.
Cisco ATA Service/Transition Extensions
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Relevant Bit of OpFlags Parameter
The relevant OpFlags parameter bit is determined by the service that is enabled or disabled, and the identity of the transition. The service types that will prompt a call to the announcement server are call waiting and call forward. Both of these services can undergo an enable transition or a disable transition. For the Cisco ATA to call the server, the applicable OpFlags bit must be set. Table 4-3 provides a mapping of each relevant OpFlags bit to its corresponding service/transition state or states.
XML Pages of Cisco ATA Information
The Cisco ATA provides XML pages that contain the following Cisco ATA information:
Current configuration
To obtain the XML configuration page, issue the following command:
http://ipaddress/dev.xmlwhere ipaddress is the IP address of the Cisco ATA whose configuration you wish to access.
This XML page is only for retrieving the configuration of a Cisco ATA; you cannot change configuration values on this page.
This XML page is password protected. You can enter the password by means of your Web browser, or you can issue the following command:
curl -d "ChangeUIPasswd=<passwd>&ChangeUIPasswd=&ChangeUIPasswd=&apply=apply"<ip addr>/dev.xmlwhere <passwd> is the Cisco ATA password and <ip addr> is the IP address of the Cisco ATA.
Current statistics
To obtain this statistics page, issue the following command:
http://ipaddress/stats.xmlwhere ipaddress is the IP address of the Cisco ATA whose statistics you wish to access.
This XML page is password protected. You can enter the password by means of your Web browser, or you can issue the following command:
curl -d "ChangeUIPasswd=<passwd>&ChangeUIPasswd=&ChangeUIPasswd=&apply=apply"<ip addr>/stats.xmlwhere <passwd> is the Cisco ATA password and <ip addr> is the IP address of the Cisco ATA.
Current service values
To obtain this service page, issue the following command:
http://ipaddress/service.xmlwhere ipaddress is the IP address of the Cisco ATA whose service values you wish to access.
This XML page is password protected. You can enter the password by means of your Web browser, or you can issue the following command:
curl -d "ChangeUIPasswd=<passwd>&ChangeUIPasswd=&ChangeUIPasswd=&apply=apply"<ip addr>/service.xmlwhere <passwd> is the Cisco ATA password and <ip addr> is the IP address of the Cisco ATA.
Complete Reference Table of all Cisco ATA SIP Services
Table 4-4 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.
Table 4-4 Configurable Features and Related Parameters
802.1Q packet tagging
Anonymity for called third party
ConnectMode, page 5-41—Bit 30
Anonymous user name support
ConnectMode, page 5-41—Bit 27
Audio compression and decompression
Audio level of FXS ports
Backup proxy configuration
Backup proxy timeout
Call forward enable/disable
ConnectMode, page 5-41—Bit 17
Call forwarding—Maximum times allowed
Call commands
Call features
Caller ID format
Call waiting
Call-waiting call ring timeout
Call-waiting hang-up alert
ConnectMode, page 5-41—Bit 25
Call-waiting state specified
Cisco Discovery Protocol
CNG tone detection
Configuration update interval
Debugging and diagnostics
NPrintf, page 5-73, TraceFlags, page 5-73, SyslogIP, page 5-74, SyslogCtrl, page 5-75
Dial plan commands
Domain name server
DNS hostname lookup
DTMF method
Encryption
Fax CED tone
Fax mode on a per-call basis
Fax pass-through
G.711 codec
Hook flash
IDs for phone lines
IP audio and signaling packets—precedence and delay
IP-like address in dial plan
Login ID
Low bit-rate codec
Mixing of tones
Network Address Translation (NAT) server—Maintain during session
NSE payload number
NTP IP address
On-hook delay
Outbound proxy
Paid features
Passwords for phone lines
Polarity
Polarity reversal before and after caller ID signal
Privacy token support for SIP diversion header
ConnectMode, page 5-41—Bit 27
Received = tag enable/disable
Receiving-audio codec preference
Redial time if line is busy
Refresh Cisco ATA using Web server
REGISTER messages
Registration removal
Reset Cisco ATA using Web server
Retransmission interval for NAT server
Retry interval if line is busy
Ringback tone—send to caller
Ring-cadence pattern
RTP media port
RTP packet size
RTP statistics
Secondary domain name server
Silence suppression setting
SIP call return
SIP proxy registrar address
SIP retransmission attempts for requests or responses
SIP proxy registration renewal
SIP registration enable/disable
SIP-request listening port
SIP session-timer settings
SessionTimer, page 5-26, SessionInterval, page 5-28, MinSessionInterval, page 5-28
Static network router probe
STUN support
TFTP file—not using internally generated name
Timing values
SigTimer, page 5-40, FeatureTimer, page 5-38, FeatureTimer2, page 5-39
Time zone offset
Tones: BusyTone, CallWaitTone
AlertTone, DialTone, ReorderTone, and RingBackTone parametersTracing
Transmitting-audio codec preference
VLAN encapsulation
VLAN mode
WAN address of NAT
Web configuration—disallowing
Posted: Tue Mar 30 16:35:27 PST 2004
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