Ready To Throw Up My Hands In Defeat

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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.

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2 thoughts on - Ready To Throw Up My Hands In Defeat

  • 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@lists.digium.com] On Behalf Of Steve Matzura Sent: Monday, May 22, 2023 10:19 AM
    To: Asterisk Users Mailing List – Non-Commercial Discussion
    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.

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  • I haven’t tried starting the daemon yet only because I wanted to verify my pjsip and extensions stuff first before I started trying to debug what I might not understand. I have a better handle on it all now. Will post the results when I try it in just a few minutes.

    Thankfully, the book does say SIP is deprecated in favor of PJSIP, so I’m on board with understanding all that. Thanks for reminding me that Google is my friend in this project, too.

    handle

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