SW: Thanks a million for the statement that I only need these two files and they can be just about empty !
David Carter: many thanks for those files which I will study Rich Adamson: That is so re-assuring! That may sound odd but its realy helpful to have the problems I am facing acknowledged and makes me feel that others really see the need for, in effect, intuitive docs to get the novice on-board. I used to write code, now I leave it to my staff, but I guess I can go there. What I am doing is evaluating * to see if we as a company should use and support it rather than just buying in Quintum boxes or whatever. No doubt many others are doing the same. As a company we write software for end-users and I insist that an average 16 year old must be able to make it work, at the basic level, without grief - it must be intuitive. OK make that "an average linux administrator" for */VOIP but again it really needs to be intuitive - but I guess I am preaching to the convereted. I would like to offer to try and do that in the wiki - but realistically I don't have the time. Still I am feeling a bit guilty now having got such solid support. Thank you. John --------------------------------------------------------- John A Coll, Director, Connection Software 391 City Road, LONDON, EC1V 1NE, UK Tel: 020 7713 8000 From outside UK Tel: +44 20 7713 8000 Fax: 020 7713 8001 Fax: +44 20 7713 8001 Email: [EMAIL PROTECTED] Web: www.csoft.co.uk PGP Public Key from keyserver -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: 03 January 2004 01:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :) > This is hard work :) I have read the Asterisk Handbook, BudgeTone User > Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages > and more. > > I am not a linux newbie but am new to Asterisk. I have failed to find any > docs that explain how to get a very very simple, minimal, system up and I am > trying to get the following to work: <snip> > I've been at this off and on for two weeks .... Linux admin and firewalls > seem trivial compared to this so I must be missing something pretty basic :) Careful, that's the wrong thing to say on this list; but, the exact same thing has been reiterated at least several thousand times (minimum) in the last few months. The underlying problem truly is that even for those of us that have been professionally involved with telephony (for years), the initial learning curve for * is far steeper then the average implementor can begin to comprehend. Please folks, let's not start the _weekly_ read the code/docs war once again; for the experienced ones that really want to click on reply, "please don't"!!!! The bottom line is that unless you can read/comprehend code rather quickly, the technical documentation does not exist in any reasonable form. Lots of very good people are trying very very hard, but the fact is that far more technical doc exists only in the code then one would expect from such an excellent application. (The subject really has been covered in very negative terms many times, if one can find it. One of the better choices for newbie research really is http://www.voip-info.org/tiki-index.php , but even this is very much a 'work-in-progress'. That's a Good Thing!!! There is also a fair number of folks on the list that are trying to earn a living via * that won't take the time to respond to even the most basic questions for obvious reasons. Their signatures will become very apparent.) Not all of the documentation problem is really related to *; there really is a lot of interpretation/advancement/research going on with SIP vendors that frequently initiate postings related to problems/comments on the list. Once you get a basic * system working, you'll find significant issues with the SIP standards in terms of NAT and many many other items. That's not putting * down, its just the nature of non-commercial internet standards. I do believe that most implementors find the /usr/src/asterisk/README.* to be helpful, and some other directories that contain sample configs (of which the directory names are so unobvious I can't find them after a couple of beers. ;) You will find that not all SIP vendors interpret the exact same standards in the same way. For those of us that have tried, software/hardware SIP phones vary dramatically in terms of interoperability with * (and other telephony apps). Some get it reasonably right, and other vendors try to advance the standards with their own interpretations. And, a few are obviously basement operations with minimal informed staff. There really are only a few _aggressive_ responders that will abrasively tell you to read the docs, but what they really mean is read the code. If that's not appropriate, then simply delete their replies; they really won't mind even a little tiny bit. It's just the nature of this list. But, keep the faith, asterisk is really very good and stable once past that initial vary-steep learning curve. Rich _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
