test-call-table
This command configures the Test Call Rules table, which allows you to test SIP signaling (setup and registration) and media (DTMF signals) of calls between a simulated phone on the device and a remote IP endpoint.
Syntax
(config-troubleshoot)# test-call test-call-table <Index> (test-call-table-<Index>)#
|
Command |
Description |
|---|---|
|
Index |
Defines the table row index. |
|
allowed-audio-coders-group-name |
Assigns an Allowed Audio Coders Group, configured in the Allowed Audio Coders Groups table, which defines only the coders that can be used for the test call. |
|
allowed-coders-mode {not-configured|preference|restriction|restriction-and-preference} |
Defines the mode of the Allowed Coders feature for the Test Call. |
|
application-type {gw|sbc} |
Application type. |
|
auto-register {disable|enable} |
Automatic register. |
|
bandwidth-profile |
Bandwidth Profile. |
|
call-duration |
Call duration in seconds (-1 for auto, 0 for infinite). |
|
call-party {called|caller} |
Test call party. |
|
called-uri |
Called URI. |
|
calls-per-second |
Calls per second. |
|
dst-address |
Destination address and optional port. |
|
dst-transport {not-configured|sctp|tcp| tls|udp} |
Destination transport type. |
|
endpoint-uri |
Endpoint URI ('user' or 'user@host'). |
|
ip-group-name |
IP Group. |
|
max-channels |
Maximum concurrent channels for session. |
|
media-security-mode {as-is|both|not-configured|rtp|srtp} |
Defines the handling of RTP and SRTP |
|
offered-audio-coders-group-name |
Assigns a Coder Group, configured in the Coder Groups table, whose coders are added to the SDP Offer in the outgoing Test Call. |
|
password |
Password for registration. |
|
play {disable|dtmf|prt} |
Playback mode. |
|
play-dtmf-method {inband|not-configured|rfc2833} |
Defines the method used by the devicefor sending DTMF digits that are played to the called party when the call is answered. |
|
play-tone-index |
Defines a tone to play from the installed PRT file. |
|
qoe-profile |
Quality of Experience (QOE) Profile. |
|
route-by {dst-address|ip-group} |
Routing method. |
|
schedule-interval |
0 disables scheduling, any positive number configures the interval between scheduled calls (in minutes). |
|
sip-interface-name |
SIP Interface. |
|
test-duration |
Test duration (minutes). |
|
test-mode {continuous|once} |
Test mode. |
|
user-name |
User name for registration. |
Command Mode
Privileged User
Example
This example partially configures a test call rule that calls endpoint URI 101 at IP address 10.13.4.12:
(config-troubleshoot)# test-call test-call-table 0 (test-call-table-0)# called-uri 101 (test-call-table-0)# route-by dst-address (test-call-table-0)# dst-address 10.13.4.12