[Asterisk-Users] parameters documentation
FELIX E SKOWRONEK
fnskoloh at msn.com
Wed Oct 12 12:34:00 MST 2005
The people who have been documenting Asterisk have been working on a book
for the last few months, it has been published by O'reilly (Asterisk-The
Future of Telephony)and is just now finding it's way into the major
bookstores, listed under Open-Source at Barns&Noble.
While it will not answer everything asterisk can do, but its glossery and
and appendix are very helpful for quick reference. If you have been
following the Asterisk Documentation Project, some of it will be old hat,
but I'm looking forward to replacing my huge stack of printouts with it.
It gives a pretty good overview of VOIP, Networking, Telephony, etc.
http://www.oreilly.com/catalog/asterisk/
>From: "Steve Totaro" <asterisk at totarotechnologies.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial
>Discussion<asterisk-users at lists.digium.com>
>To: "Asterisk Users Mailing List - Non-Commercial
>Discussion"<asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] parameters documentation
>Date: Wed, 12 Oct 2005 11:17:01 -0400
>
>There is plenty of documentation online for both the 3com and *. You have
>to have good search skills I guess.
>
>3com has the best knowledge base I have seen.
>http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
>help.
>
>I think you may need to learn some basic networking before learning
>asterisk. NAT is a very basic concept in networking as well as ports such
>as 5060 (standard port for SIP).
>
>There is a very steep learning curve for asterisk and networking in
>general.
>If you want to learn it then you need to dig into the wiki and read all the
>posts that come across the user's list (well maybe not all of them).
>
>There are plenty of consultants that you can hire if you are not up to it.
>
>
>----- Original Message -----
>From: <asterisk at frameweb.it>
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
><asterisk-users at lists.digium.com>
>Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
><asterisk-users at lists.digium.com>;
><asterisk-users-bounces at lists.digium.com>
>Sent: Wednesday, October 12, 2005 10:34 AM
>Subject: Re: [Asterisk-Users] parameters documentation
>
>
> > I come from a NBX100
> > No documentation available.
> > 1 day it starts saying: "syslog full" and voicemail stop working
> > No one was able to tell me what was the meaning of that alert
> > .
> > 3COM NBX anyway is a good product, but the price is too high, especially
>4
> > years ago, and especially the price of the telephone is very high.
> >
> > Andrea
> >
> >
> >
> >
> >
> > "asterisk"
> > <asterisk at totarot
> > echnologies.com>
>To
> > Sent by: "Asterisk Users Mailing List -
> > asterisk-users-bo Non-Commercial Discussion"
> > unces at lists.digiu <asterisk-users at lists.digium.com>
> > m.com
>cc
> >
> >
>Subject
> > 13/10/2005 16.13 Re: [Asterisk-Users] parameters
> > documentation
> >
> > Please respond to
> > Asterisk Users
> > Mailing List -
> > Non-Commercial
> > Discussion
> > <asterisk-users at l
> > ists.digium.com>
> >
> >
> >
> >
> >
> >
> > "> I really hope this project will be implemented, without documentation
> > evrything is too hard"
> >
> > Not for the thousands of people that have figured it out.
> >
> > 3Com NBX might be more your speed and plenty of documentation.
> >
> >
> >
> > > Really strange answer. I am non used to search on playboy.com.
> > >
> > > Anyway, if you try to search
> > > insecure=very
> > > on www.voip-info.org, you find 742 links , a bit more for me. (I just
> > want
> > > to know what it means)
> > >
> > > Moreovere, the first 20 links are non accessible at all
> > >
> > >
> >
>http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecure&diff=6
> >
> > >
> > > they speak about tiki-pagehistory.php, which appears not to exist.
> > >
> > > no other comments about this.
> > > ************************************************************
> > >
> > > I know about one project , "asterisk documentation project"
> > >
> > > http://www.asteriskdocs.org
> > >
> > > in its home page, the first line is
> > >
> > >
> > >
> > >
> > >
> > > Great software needs great documentation.
> > >
> > >
> > >
> > >
> > > I really hope this project will be implemented, without documentation
> > > evrything is too hard
> > >
> > > Andrea
> > >
> > >
> > >
> > >
> > > "Steve Totaro"
> > > <asterisk at totarot
> > > echnologies.com>
> > To
> > > Sent by: "Asterisk Users Mailing List -
> > > asterisk-users-bo Non-Commercial Discussion"
> > > unces at lists.digiu
><asterisk-users at lists.digium.com>
> > > m.com
> > cc
> > >
> > >
> > Subject
> > > 12/10/2005 14.53 Re: [Asterisk-Users] parameters
> > > documentation
> > >
> > > Please respond to
> > > Asterisk Users
> > > Mailing List -
> > > Non-Commercial
> > > Discussion
> > > <asterisk-users at l
> > > ists.digium.com>
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > www.voip-info.org
> > >
> > > > Another trivial question:
> > > >
> > > > Is there a "place" where all the parameters are documented ?
> > > > In example (my example!) I would like to know the meaning of a lot
>of
> > > > parameter that can be used in sip.conf,
> > > >
> > > > A lot of these keywords are intuitive keywords (i.e.
> > > NAT=YES/NO;PORT=5060;
> > > > context=xxxxx) but other are not (at least for me)
> > > > i.e.:
> > > >
> > > > type = peer, friend
> > > > insecure=very
> > > > host=dynamic
> > > >
> > > > and so on.
> > > >
> > > > At last, my need is:
> > > >
> > > > Accept a non-registerd sip-strem from a well known ip address (and
>only
> > > > from that ip address....)
> > > >
> > > > I tried to add a
> > > >
> > > > [testsip]
> > > > ;username=testsip
> > > > type=friend
> > > > ;secret=testsip
> > > > qualify=no
> > > > port=5060
> > > > nat=no
> > > > host=x.y.z.w
> > > > dtmfmode=rfc2833
> > > > context=from-internal
> > > > canreinvite=no
> > > > callerid="test sip " <testsip>
> > > >
> > > > that would work if the sip would be registered. But the SIP client
>is
> > not
> > > > able to register.
> > > >
> > > > I solved using the
> > > > context = from-sip-external ; Send unknown SIP callers to this
>context
> > > >
> > > > and it works, but I have no more the control about who is sending me
> > SIP
> > > > stream (anybody now can use my asterisk box...)
> > > >
> > > > any help will be greatly appreciated
> > > >
> > > > Andrea
> > >
> > > _______________________________________________
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> > >
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> > >
> > >
> > > --
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> >
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>
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