Unfortunately, Asterisk 1.6.2 was completed at the end of April 2012, and it seems that this feature is still not supported, and using a sound card with Asterisk is not the most common use case.
Installing Asterisk from the source is quite simple and also pretty clean ( make uninstall will clean it up enough). I highly recommend it, as it allows you to get ahead of the game (with protection and release of functions). Try with ./configure and then make menuselect (you'll need ncurses libraries) for a really nice build interface.
When trying to test outgoing calls on your SIP trunk (to check if it is connected), I would recommend using the channel originate function in the CLI.
For reference:
asterisk*CLI> core show help channel originate
The specific device and parameter string using your setup:
asterisk*CLI> channel originate SIP/flowroute/00359891505054 application Playback tt-monkeys
Note. This will play the sound of screaming monkeys to the called party!
In addition, if you are a smartphone user, it is quite convenient to connect a softphone as an extension to check your trunk lines, new dialplans, etc. (Although I will always perform my initial tests with channel originate )
Finally, but no less important ... I understand that this is a test scenario, but ... How do you do it; I would avoid directly dialing your ITSP with a dialed extension from the context [default] . This could, if you did not configure the other configuration properly, leave you open to fraud using the fact that when you try to call the default for an asterisk, this context is used by default if the context is unknown.