|
|
Max Forward |
SIP Max Forward value, which can range from 1 to 255. Defaults to 70. |
Max Redirection |
Number of times an invite can be redirected to avoid an infinite loop. Defaults to 5. |
Max Auth |
Maximum number of times (from 0 to 255) a request may be challenged. Defaults to 2. |
SIP User Agent Name |
Used in outbound REGISTER requests. Defaults to $VERSION. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed. |
SIP Server Name |
Server header used in responses to inbound responses. Defaults to $VERSION. |
SIP Reg User Agent Name |
User-Agent name to be used in a REGISTER request. If this is not specified, the <SIP User Agent Name> is also used for the REGISTER request. Defaults to blank. |
SIP Accept Language |
Accept-Language header used. To access, click the SIP tab, and fill in the SIP Accept Language field. There is no default (this indicates SPA9000 does not include this header). If empty, the header is not included. |
DTMF Relay MIME Type |
MIME Type used in a SIP INFO message to signal a DTMF event. This field must match that of the Service Provider. Defaults to application/dtmf-relay. |
Hook Flash MIME Type |
MIME Type used in a SIP INFO message to signal a hook flash event. The default is application/hook-flash. |
Remove Last Reg |
Lets you remove the last registration before registering a new one if the value is different. Select yes or no from the drop-down menu. Defaults to no. |
Use Compact Header |
Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the phone uses compact SIP headers in outbound SIP messages. If set to no, the phone uses normal SIP headers. If inbound SIP requests contain compact headers, the phone reuses the same compact headers when generating the response regardless the settings of the <Use Compact Header> parameter. If inbound SIP requests contain normal headers, the phone substitutes those headers with compact headers (if defined by RFC-261) if <Use Compact Header> parameter is set to yes. Default: no |
Escape Display Name |
Lets you keep the Display Name private. Select yes if you want the IP phone to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside the pair of double quotes. Otherwise, select no. Defaults to yes. |
Escape Special Character |
Normally, the IP phone sends a “%23” (escape) as part of the message when the special character # is included in the SIP INVITE message. This can cause problem s with some telephony servers that need to receive the # character. When this parameter is set to yes, the phone sends a “%23” (escape) as part of the message when the # is included. When set to no, the # is sent directly and the escape (%23) is not used. Defaults to yes. |
SIP-B Enable |
Enables SIP for Business (supports Sylantro call flows) call features. |
Talk Package |
Enables support for the BroadSoft Talk Package, which enables a user to answer or resume a call by clicking a button in an external application. |
Hold Package |
Enables support for the BroadSoft Hold Package, which enables a user to place a call on hold by clicking a button in an external application. |
Conference Package |
Enables support for the BroadSoft Conference Package, which enables a user to start a conference by clicking a button in an external application. |
Notify Conference |
If enabled, the unit will send out a NOTIFY with event=conference when starting a conference. |
RFC 2543 Call Hold |
If set to yes, unit will include c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either case. Defaults to yes. |
Random REG CID On Reboot |
If set to yes, the Cisco IP phone uses a different random call-ID for registration after the next software reboot. If set to no, the Cisco IP phone tries to use the same call-ID for registration after the next software reboot. The Cisco IP phone always uses a new random Call-ID for registration after a power-cycle, regardless of this setting. Defaults to no. |
Mark All AVT packets |
If set to yes, all audio video transport (AVT) tone packets (encoded for redundancy) have the marker bit set. If set to no, only the first packet has the marker bit set for each DTMF event. Defaults to yes. |
SIP TCP Port Min |
Specifies the lowest TCP port number that can be used for SIP sessions. Defaults to 5060. |
SIP TCP Port Max |
Specifies the highest TCP port number that can be used for SIP sessions. Defaults to 5080. |
CTI Enable |
The CTI interface allows a third-party application to control and monitor the state of a phone that has registered with the Cisco SPA9000. With this interface, an application can control a phone to initiate an outgoing call or answer an incoming call with a mouse click from a PC. |
Caller ID Header |
Provides the option to take the caller ID from PAID-RPID-FROM, P-ASSERTEDIDENTITY, REMOTE-PARTY-ID, or FROM header. |
SRTP Method |
Selects the method to use for SRTP. Two choices are available:
- x-sipura—legacy SRPT method
- s-descriptor—new method compliant with RFC-3711 and RFC-4568
The default value is "x-sipura.” Note Not applicable to Cisco WIP310. |
Hold Target Before REFER |
Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended call transfer (where the transfer target has answered). Default value is "no,” where the call leg is not held. Note Not applicable to Cisco WIP310. |
Dialog SDP Enable |
When enabled and the Notify message body is too big causing fragmentation, the Notify message xml dialog is simplified; Session Description Protocol (SDP) is not included in the dialog xml content. |
Keep Referee When REFER Failed |
Set this parameter to yes to configure the phone to immediately handle NOTIFY sipfrag messages. You can also configure this parameter in the configuration file:
<Keep_Referee_When_REFER_Failed ua="na">Yes
</Keep_Referee_When_REFER_Failed>
|
Display Diversion Info |
Set to yes to cause the IP phone to parse and display the Diversion header in the incoming INVITE messages. The phone displays the information from the first Diversion header; if there are multiple Diversion headers, the others are ignored. The Diversion header follows the definition from RFC 5806. For example, for this diversion header in the INVITE messages:
Diversion: <sip:WeSellFlowers@p4.isp.com>;reason=time-of-day
The screen should display:
The reason field in the header is ignored. If the header field is not properly formed, Diversion header (all) will be ignored. If the privacy field is not set or is set to off, only the URI is displayed due to screen size limitation. If the privacy field is set to full, “anonymous” is displayed. If the privacy field is set to name, only the display name is displayed. |
Display Anonymous From Header |
Set to yes to show the caller ID from the SIP INVITE message “From” header, even if the call is an anonymous call. When the parameter is set to no, the phone displays "Anonymous Caller" as the caller ID. |
Disable Local Name To Header |
The options are No and Yes:
- If No is selected, no changes are made. The default value is No.
- If Yes is selected, the following happens:
– Disables the display name in “Directory” and “Call History” in the “To” header during an outgoing call. – Ignores the CLID in the “UPDATE” message. – Redial list is populated based on 18x or 200 OK PAID header with or without Display Name. Note This field is supported in Firmware Release 7.6.2 and later. |