diff --git a/apps/app_dial.c b/apps/app_dial.c index 7d7a4594f9..ee77551143 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -613,12 +613,31 @@ This is the status of the call - - - - - - + + Either the dialed peer exists but is not currently reachable, e.g. + endpoint is not registered, or an attempt was made to call a + nonexistent location, e.g. nonexistent DNS hostname. + + + Channel or switching congestion occured when routing the call. + This can occur if there is a slow or no response from the remote end. + + + Called party did not answer. + + + The called party was busy or indicated a busy status. + Note that some SIP devices will respond with 486 Busy if their Do Not Disturb + modes are active. In this case, you can use DEVICE_STATUS to check if the + endpoint is actually in use, if needed. + + + The call was answered. + Any other result implicitly indicates the call was not answered. + + + Dial was cancelled before call was answered or reached some other terminating event. + For the Privacy and Screening Modes. Will be set if the called party chooses to send the calling party to the 'Go Away' script. @@ -627,7 +646,9 @@ For the Privacy and Screening Modes. Will be set if the called party chooses to send the calling party to the 'torture' script. - + + Dial failed due to invalid syntax. + @@ -3569,4 +3590,4 @@ AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Dialing Application", .load = load_module, .unload = unload_module, .requires = "ccss", -); \ No newline at end of file +);