Please explain your trunks. It appears you have one called 9844804673 and one called Hope_For_Felons_Inc . Your Outbound Route tried both and they both failed. Are these with different providers or different accounts? However, we can’t see why, because pjsip logger got turned off, or was never on.
At the Asterisk command prompt (not a shell prompt) type pjsip set logger on
you should see PJSIP Logging enabled
then make your test call and paste a new log. Note that when you Apply Config (or otherwise restart or reload Asterisk), pjsip logger gets turned back off.
Neither of your pastes show an outbound call attempt. They both show attempted attacks, which appear not to have done any harm, but have caused a lot ot needless logging, which makes it difficult to troubleshoot real problems. It appears that (in Asterisk SIP Settings), you have Allow Anonymous Inbound SIP Calls and/or Allow SIP Guests turned on, which cause these attacks to be processed. I am guessing that you have UDP port 5060 open to the outside world. Either your router/firewall or FreePBX firewall should be set to block all such requests except from your trunking provider(s) and any external extensions you may have.
Log starts too late to be of any use. The call has already failed. Also there is no PJSIP logging, but that might be because it ends before the incoming side completes.
No it would not work. It would give a fast busy tone and would not send to freepbx. I have came to the conclusion that for some reason when we do a simple dial plan that should work with both freepbx and HT802 it will not. All incoming calls work great I just can’t make calls. After researching I have not seen anyone that has been successful in setting up the HT802 with freepbx. I just purchased this one from Amazon so I can easily send this one back and exchange for something different. I would like to know if there is a ATA that has been successfully setup with freepbx I’m willing to give it a try. It looks to me that the dial plan is sent to the Freepbx so both dial plans have to match if not it’s simply not going to work. I have been working on this for three months paying for the trunk without a positive outcome. I have thought about purchasing a VoIP phone that was compatible with Freepbx and roll with it. Is there any free versions of soft phones for Android and Windows that I could use while I’m trying to set this up?
I will run that again. The setup is Hope For Felons is the inbound and the 9844804673 I use for outgoing. But I did have 9844804673 as the main outgoing and the other as second but I took the Hope For Felons out.
I feel that the Grand team H802 may not be comfortable with freepbx as far as outbound because I have researched this channel and saw the same issues dated over 10 years ago. I’m all about trying to receive help. I just don’t know anyone in this channel that owns the device lol. Anyway I will run that and do some test calls and post them in a few.
371 [2025-06-01 11:06:47] VERBOSE[99291][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:18] Set("PJSIP/4-00000004", "OUTNUM={1[2-9]xx[2-9]xxxxxx}13367890123") in new stack
Something is very wrong with your outbound route! You have a pattern where you should have a literal. That suggests you have put something in the prepend column that should be in one of the other columns.
In fact, it contains characters (“{”, and “}”) which should not be anywhere near a FreePBX outbound route definition.
This is not an HT802 issues; it is purely a FreePBX configuration error.
At this point, you probably need to ask the provider why they don’t like you, although I would note that your From user is missing, and many providers use that to identify the account.
From user defaults to the caller ID, but can be overridden
You do set a caller ID, but then immediately destroy it:
933 [2025-06-01 15:09:17] VERBOSE[248313][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:30] ExecIf("PJSIP/4-0000000e", "1?Set(CALLERID(all)=9844804673)") in new stack
934 [2025-06-01 15:09:17] VERBOSE[248313][C-00000009] pbx.c: Executing [s@macro-outbound-callerid:31] ExecIf("PJSIP/4-0000000e", "1?Set(CALLERID(all)=Hope For Felons Inc )") in new stack
I’m not actually sure that even the first line sets the caller ID; it might just set the caller name.
Several problems with the Nextiva trunk: On the pjsip Settings tab, Advanced:
Set From User to the same value you have in Username.
Set From Domain to the same value you have in SIP Server.
After Submit and Apply Config, turn pjsip logger back on.
Make a test call to 18004377950 (note inital 1) , report what you hear and paste a new log.
Thank you. That fixed the problem. I can now make calls yay. I have been trying to figure this out for a long time. Thanks again. Do you have suggestions on setting op a conference and how to do that? I have 3 numbers: my main number, a fax number, and one for conference.