asterisk users - May 2023 - Ready to throw up my hands in defeat

I am not comfortable with admitting this on a public userlist [;-)] but 
after over forty years in software development and manual-reading and 
-interpretation, I've finally hit one that I can't get past.


I've mention previously that I worked with Asterisk in older days--like 
in around 2003--and never had any trouble understanding what to do and 
how to do it in order to make it work. I am attempting to build what's 
probably the world's most basic system--one incoming trunk from a DID 
provider going to one internal extension that answers, plays a couple 
things, and possibly takes a message. I'd also like to add two 
extensions with real physical endpoints--phones--one local, one remote. 
I think I can manage that part. It's the initial SIP stuff that's making
me dizzy.


The book I am now reading--"Asterisk, the Definitive Guide" by Madsen,
Bryant and Meggelin for Asterisk version 16-- assumes I have built an 
implementation from source, and that includes SQL. There are tons of 
references to SQL databases in the book which I understand, but having 
installed Asterisk from a distribution package, that component is not 
part of the installation, so I am presumably expected to supply the 
information by manually entering it into configuration files. I'm OK 
with doing that, too. The part I'm having trouble with is that the 
samples in the configuration files, particularly pjsip.conf, offer 
several choices for some of the stanzas, like all the things defining 
trunks and endpoints, and that's where I'm losing it. The book makes it 
sound and look so easy--add a couple records to a couple SQL tables 
according to your instruments and DID providers, and it probably works 
just that smoothly and easily. But how does one make these choices when 
one has to manually edit these configurations and choose the one that at 
least halfway looks like the SQL stuff in the book?


I think I need a little hand-holding and am willing to buy some from 
someone who has the time and inclination to provide it. I'm a fast 
learner, I record all such sessions, and I'm sure I can get what I need 
in a couple hours, most likely less. if you're interested, or know 
someone who is, please contact me off-list, with my eternal thanks in 
advance.

There are lots of little tweaks/adjustments overlooked in most guides/books. 
The examples work most of the time, but even a small difference in your
environment might break them.

I'm pretty sure the list will be able to answer questions to help you figure
it out.  If you break down your current problem into the basic step/task and
explain what's not working then you'll likely get a good explanation.

If you're not sure where to start, just add one physical phone  and a
screaming monkeys entry in the dialplan (lots of examples out there).  If
that' doesn't work, post the CLI output with verbose turned up.

In general stay away from realtime (I assume that is the SQL reference)


-----Original Message-----
From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On
Behalf Of Steve Matzura
Sent: Monday, May 22, 2023 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
Subject: [asterisk-users] Ready to throw up my hands in defeat

I am not comfortable with admitting this on a public userlist [;-)] but after
over forty years in software development and manual-reading and -interpretation,
I've finally hit one that I can't get past.


I've mention previously that I worked with Asterisk in older days--like in
around 2003--and never had any trouble understanding what to do and how to do it
in order to make it work. I am attempting to build what's probably the
world's most basic system--one incoming trunk from a DID provider going to
one internal extension that answers, plays a couple things, and possibly takes a
message. I'd also like to add two extensions with real physical
endpoints--phones--one local, one remote.
I think I can manage that part. It's the initial SIP stuff that's making
me dizzy.


The book I am now reading--"Asterisk, the Definitive Guide" by Madsen,
Bryant and Meggelin for Asterisk version 16-- assumes I have built an
implementation from source, and that includes SQL. There are tons of references
to SQL databases in the book which I understand, but having installed Asterisk
from a distribution package, that component is not part of the installation, so
I am presumably expected to supply the information by manually entering it into
configuration files. I'm OK with doing that, too. The part I'm having
trouble with is that the samples in the configuration files, particularly
pjsip.conf, offer several choices for some of the stanzas, like all the things
defining trunks and endpoints, and that's where I'm losing it. The book
makes it sound and look so easy--add a couple records to a couple SQL tables
according to your instruments and DID providers, and it probably works just that
smoothly and easily. But how does one make these choices when one has to
manually edit these configurations and choose the one that at least halfway
looks like the SQL stuff in the book?


I think I need a little hand-holding and am willing to buy some from someone who
has the time and inclination to provide it. I'm a fast learner, I record all
such sessions, and I'm sure I can get what I need in a couple hours, most
likely less. if you're interested, or know someone who is, please contact me
off-list, with my eternal thanks in advance.


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users