VoIP server settings:
To create new VoIP server settings, click the Create button.
To create a redundant VOIP server, select a VOIP server and click Create Redundant. This will create a redundant VOIP server for the selected server.
To modify a VOIP server, select a server and click Modify.
To delete a server, select a server and click Delete.
To create an Emergency Standalone (ESA) fallback server, select a server then click Create ESA Fallback.
Note that if you already have a primary VOIP server using the megaco protocol, then you only need to create additional VOIP server with sip protocol for the ESA fallback and the SIP dialplans.
To remove the displayed settings, click the check box next to the desired settings, and click the Delete button.
Appl Index |
Index number for this entry. | ||||||||||||||||||
Server Address Index |
Index of VOIP server. | ||||||||||||||||||
Server Address Type |
The type of SIP address used. | ||||||||||||||||||
Server Address |
Specifies the address of a VOIP server this user agent will use to proxy/redirect calls. | ||||||||||||||||||
UDP Port Number |
The UDP Port number for the MGCP Gateway. The default 2427 is the IANA Registered MGCP Gateway port. | ||||||||||||||||||
Server Type |
Identifies the VOIP server type. The default is generic. | ||||||||||||||||||
Server Protocol |
The signalling protocol used for VOIP. Choices are SIP (Session Initiation Protocol) and MGCP (Media Gateway Control Protocol). | ||||||||||||||||||
Send Call Proceed Tone |
Indicates whether, on call originations destined to the PSTN, the device should play a call proceeding tone while waiting for ring-back tone from the switch. | ||||||||||||||||||
RTCP Enabled |
Indicates whether the Real-Time Control Protocol (RTCP) is enabled for the device. | ||||||||||||||||||
RTCP Packet Interval (msec) |
Real-Time Control Protocol (RTCP) packet interval, in milleseconds. Range is 2500 to 10000 ms. Default is 5000 ms. | ||||||||||||||||||
Inter-Digit Timeout (sec) |
The duration, in seconds, that the device will wait after each digit is entered before assuming the user has finished entering digits. | ||||||||||||||||||
IP ToS |
The ToS value that is set in the ip header for voice of ip traffic.
Default: 0 | ||||||||||||||||||
Domain Name |
The Universal Resource Identifier (URI) of the system. | ||||||||||||||||||
Expires Invite (h:m:s) |
For user agent clients, this value is inserted into the Expires header. For proxy servers, if a received request contained an Expires header, then the value in the header takes precedence. | ||||||||||||||||||
Expires Register (h:m:s) |
For user agent clients, this value is inserted into the Expires header. For registrar servers, if a received request contained an Expires header, then the value in the header takes precedence. | ||||||||||||||||||
Server Header Method |
This convention is a bit map. Each bit represents a SIP method where the Expires header should be included. If a bit has value 1, then the requests corresponding to that SIP method must include an Expires header line. If a bit has value 0, then the Expires header line will not be added. Combinations of bits can be set when the Expires header line is required in multiple SIP methods. bit 0 : INVITE method bit 1 : REGISTER Method | ||||||||||||||||||
Session Timer |
Support session timers. on: Session timers are supported, and SIP messages will have a 'Supported: timer' header. off: Session timers are NOT supported, and SIP messages will NOT have a 'Supported: timer' header. Default is off. | ||||||||||||||||||
Session Expiration Timeout (sec) |
The time at which an element (proxy, UAC, UAS) will consider the SIP session timed out if no successful session refresh transaction occurs beforehand. | ||||||||||||||||||
Min Session Exp Timeout (sec) |
Because of the processing load of mid-dialog requests, all elements (proxy, UAC, UAS) can have configured minimum value for the session interval that they are willing to accept. | ||||||||||||||||||
Caller Request Timer |
This field specifies the caller's action yes: Caller (UAC) will request for timer when making outbound calls. no: Caller (UAC) will NOT request for timer when making outbound calls. Default is no. | ||||||||||||||||||
Callee Request Timer |
When caller supports timer but did not request one this field specifies the callee's action. yes: Callee (UAS) will request for timer. no : Callee (UAS) will NOT request for timer. Default is no. | ||||||||||||||||||
Caller Specify Refresher |
Caller specifies the refresher: UAC : Caller specifies itself (UAC) as the refresher. UAS : Caller specifies the callee (UAS) as the refresher. Omit: Caller does NOT explicitly specify the refresher. Default is omit. | ||||||||||||||||||
Callee Specify Refresher |
When UAC did not specify refresher tag, callee specifies the refresher: UAC : Callee specifies the caller (UAC) as the refresher. UAS : Callee specifies itself (UAS) as the refresher. Default is UAC. | ||||||||||||||||||
DTMF Mode |
Indicates whether dtmf digits are to be passed as codec. Voice packets or special separate packets (rfc2833). Default is rfc2833. | ||||||||||||||||||
RTP TermId Syntax |
Describes the Ephemeral term ID syntax for the switch. The RTP syntax varies from switch to switch.But the string should contain at least one %eid format, eg <string>%eid<string>. This field is supported only for the Megaco protocol for all other protocols this field will be initialized to NULL. If the protocol is megaco and the field is NULL, Ephemeral term id will be updated to default syntax of the server Id given. If the server Id is not provisioned with the default id then field will be updated to NULL. | ||||||||||||||||||
RTP DSCP |
Indicates DSCP value for the VOIP server RTP traffic. The value selected indicates how the packets are marked | ||||||||||||||||||
Signaling DSCP |
Indicates DSCP value for the VOIP server signaling traffic. The value selected indicates how the packets are marked |
February 24, 2012