[Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal

Leif Madsen leif at radiokaos.com
Mon Aug 4 16:28:07 MST 2003


> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of John Todd
> Sent: August 4, 2003 6:35 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal
> If not, then you're
> missing an important part of the debug (and perhaps having a
> conceptual problem with how call flow works.)

What are some of the best documents a new person to Asterisk should read
in order to get a complete understanding of how the call flow works and
how they are related to how the scripts are formed.  I am pretty sure I
understand, but just want to make sure I've read everything in that
regard.

Thanks.




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