[Web-meetme-users] Configuration example for 'INVITE-Feature'

[<Sm...@gm...>]

Hello everybody,

I'm looking for an example how to configure the 'invite'-feature. I tried to put in the CHAN_TYPE, OUT_CONTEXT and OUT_PEER values but everytime I click the CALL button my asterisk crashs with the following messages:

<-- snip -->
*CLI> Nov 24 16:40:21 WARNING[24830]: cdr.c:550 ast_cdr_disposition: Cause not handled
Nov 24 16:40:21 WARNING[24830]: cdr.c:550 ast_cdr_disposition: Cause not handled
*CLI> Ouch ... error while writing audio data: : Broken pipe

Disconnected from Asterisk server
[1]+  Segmentation fault      (core dumped) asterisk -gvvvvvvvvv
<-- snap -->

What is exactly required in these variables?
Any help or explanation is highly appriciated!

Thanks a lot in advance,
Martin
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