This returns to Asterisk CLI
VPS.IP > VPS IP address
MY.IP > my local IP address
[2023-03-13 16:40:33] NOTICE[24347] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:VOIP-main@VPS.IP>' failed for 'MY.IP:18293' (callid: 5lBKi7zeO6qdR4KcGtn1HQ..) - No matching endpoint found
[2023-03-13 16:40:33] NOTICE[24347] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:VOIP-main@VPS.IP>' failed for 'MY.IP:18293' (callid: 5lBKi7zeO6qdR4KcGtn1HQ..) - No matching endpoint found
[2023-03-13 16:40:33] NOTICE[24347] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:VOIP-main@VPS.IP>' failed for 'MY.IP:18293' (callid: 5lBKi7zeO6qdR4KcGtn1HQ..) - Failed to authenticate
I really don’t know what to do about this as nothing I tried has worked so far
Now I ran across another issue, I don’t know how to connect to it using Zoiper. There is no password defined, I don’t even know if I’m trying to register (from my Zoiper) under the right username. https://i.imgur.com/djqn1sC.png
^^ I assume that route is used for connecting to it, I’m really just floating around, don’t know what I’m really doing
Thank you in advance, it’s been really helpful so far
Another update: I created an extension and routes for outbound calls. Connected to it using Zoiper, no issues so far.
Upon trying to dial a number, it got rejected with error Decline (code: 603)
Asterisk CLI returned this error: ERROR[32255] loader.c: Error loading module 'app_macro': /usr/lib/asterisk/modules/app_macro.so: cannot open shared object file: No such file or directory
Insufficient logging. However providers will often use 603 if you violate policy rules, e.g. not using your assigned caller ID.
app_macro is deprecated, and will not be built by default. You should not try to load it unless your dialplan uses it, and you have selected it to be built.
Like chan_sip, it is no longer in the development branch of the source code, so will not be in this year’s release of Asterisk.
Note to Asterisk developers, I wonder if you have underestimated the number of users copying and pasting out of date (and often buggy) tutorials, who are going to find they can no longer just copy an example.
I see, not really sure what to do then, I followed instructions as shown on documentations you sent me.
PJSIP is being used. Is it possible somehow not to use said app_macro ?
To not use it, simply don’t use it. Nothing in Asterisk will use it unless you explicitly request its use.
However in this case, I think it is because you have a modules.conf that is explicitly trying to load it, in which case you should remove it from modules.conf.
Note that FreePBX makes heavy use of it, the video you referenced was for FreePBX. If you are using FreePBX, you should use their forum for support. (The video also includes username, which is an obsolete name for something that doesn’t do what people are assuming it does.)
The instructions I provided don’t cover the user of app_macro, one way or the other. It is not relevant to the chan_sip / chan_pjsip difference. Also the examples in that documentation are examples; they may need to be modified, by reference to the full list of options, and an understanding of what the provider actually requires. They will generally work with a provider that does things in a straightforward way and particular ones expecting to be used by a phone, rather than a PABX.
