[asterisk-users] Call interruption

[Yuan LIU]

>From: "Andre Wangler" <awangler at ee.ethz.ch>
>Date: Fri, 4 May 2007 07:35:38 +0200
>
>Hello all
>
>Could someone tell me what happens with running calls when reloading the 
>whole asterisk config files? I think SIP-calls are not

Nothing.  All calls are maintained according to documentation.

Yuan Liu

>interrupted because of the protocol architecture (signalling vs. media) but 
>what's with other kind of calls like h323 or over analogue interfaces? are 
>they interrupted?
>I'm quite new with asterisk, so excuse this probably trivial question...
>
>Andre