[Noob] Clients can’t call each other


I’m a complete noob and just wanted to explore voip with asterisk. I have Windows server 2022 (ADDS, DNS, DHCP), 2 clients on Windows 10 pro. I added my asterisk server (ubuntu) in the AD and just wanted to allow my users to call each other with zoiper. So I followed a YouTube tutorial but it doesn’t work for me and seems to work for everyone else lol.

When I type this command :
channel originate SIP/7001 extension 7002

I receive a call and when I answer a voice says “I’m sorry, this is not a valid extension”

Please help me, l’d really like for this to work and wanted to present this solution to my teacher but when I see that I can’t do something that looks so simple… it’s discouraging.

Thanks in advance, I posted my conf files here.

We’d need to see the tutorial. Either it contains a quite complex dialplan, or it installs something (possibly FreePBX) which provides a lot of dialplan.

Also chan_sip is obsolete and unsupported.

When asking questions, you should detail what you have done and provide logs showing, in detail, what happened.

Technically, to meet your requirement, your diaplan only needs to consist of two lines, the default context name and one for the called extension, which invokes Dial, assuming there are, valid, type=peer entries for 7001 and 7002, although you should really move to chan_pjsip.

Also note that you are not using client and server in the sense in which SIP uses them. One consequence of that is that most SIP phones can call another SIP phone without needing a PABX, and your question could be read as how to do that.

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Actually. because you are using CLI Originate, both phones are SIP servers, and Asterisk is the client for both of them. Neither of your phones is calling anybody.

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Thanks for your explanations and for answering so quickly!
Oh I kind of understand a little bit now. Also, should I install pjsip or is it a file that I have to create? Sorry for asking stupid questions, I’m really lost…

Here is the tutorial I followed : https://youtu.be/rtHFdhCm434?si=6AGyBfhzteXDLyXq


Not a good start! (Although it is possible that our soft phones don’t allow direct SIP calls.)

The sip.conf is not too bad, although local networks is useless here and it has some very common bad choices of settings. The worst thing is that it fails to limit the possible codecs, which can break some systems. As I said you should forget about it if you are going to continue using Asterisk.

extensions.conf uses Answer unnecessarily, and, in the real world, you would probably want to produce different messages for different reasons for the call failing.

I think, though, that your real problem is that Originate’s default context is “default”, but your extensions are in “internal”. Although, if that is the case, it is the phone that would be generating the announcement. The tutorial doesn’t actually cover the use of Originate!