Cant make outbound calls. Inbound calls work fine

Hello,
I bought a did number from voip.ms. I have it registered and shows its working on that site. I can recieve calls no issue. I am using microsip to make calls out and I cant get it to work.
I hear

The number you have dialed is not in service for ANY nuymber I call adding a 1 +1 or regular number

Here is my log.

[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/206.189.202.87-0000001c", "0?Set(CDR(recordingfile)=)") in new stack
[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] pbx.c: Executing [s@macro-hangupcall:4] NoOp("SIP/206.189.202.87-0000001c", " montior file= ") in new stack
[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] pbx.c: Executing [s@macro-hangupcall:5] GotoIf("SIP/206.189.202.87-0000001c", "1?skipagi") in new stack
[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] pbx.c: Executing [s@macro-hangupcall:7] Hangup("SIP/206.189.202.87-0000001c", "") in new stack
[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/206.189.202.87-0000001c' in macro 'hangupcall'
[2024-06-13 16:38:55] VERBOSE[3512][C-00000039] pbx.c: Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/206.189.202.87-0000001c'
[2024-06-13 16:38:58] VERBOSE[1751][C-0000003a] netsock2.c: Using SIP RTP TOS bits 184
[2024-06-13 16:38:58] VERBOSE[1751][C-0000003a] netsock2.c: Using SIP RTP CoS mark 5
[2024-06-13 16:38:58] WARNING[1751][C-0000003a] res_format_attr_siren7.c: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
[2024-06-13 16:38:58] WARNING[1751][C-0000003a] res_format_attr_siren14.c: Got siren14 offer at 24000 bps, but only 48000 bps supported; ignoring.
[2024-06-13 16:38:58] WARNING[1751][C-0000003a] res_format_attr_siren14.c: Got siren14 offer at 32000 bps, but only 48000 bps supported; ignoring.
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [7743342638@from-sip-external:1] NoOp("SIP/206.189.202.87-0000001d", "Received incoming SIP connection from unknown peer to 7743342638") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [7743342638@from-sip-external:2] Set("SIP/206.189.202.87-0000001d", "DID=7743342638") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [7743342638@from-sip-external:3] Goto("SIP/206.189.202.87-0000001d", "s,1") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx_builtins.c: Goto (from-sip-external,s,1)
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-sip-external:1] GotoIf("SIP/206.189.202.87-0000001d", "1?setlanguage:checkanon") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx_builtins.c: Goto (from-sip-external,s,2)
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-sip-external:2] Set("SIP/206.189.202.87-0000001d", "CHANNEL(language)=en") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-sip-external:3] GotoIf("SIP/206.189.202.87-0000001d", "0?noanonymous") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-sip-external:4] Goto("SIP/206.189.202.87-0000001d", "from-trunk,7743342638,1") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx_builtins.c: Goto (from-trunk,7743342638,1)
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [7743342638@from-trunk:1] Set("SIP/206.189.202.87-0000001d", "__FROM_DID=7743342638") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [7743342638@from-trunk:2] NoOp("SIP/206.189.202.87-0000001d", "Received an unknown call with DID set to 7743342638") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [7743342638@from-trunk:3] Goto("SIP/206.189.202.87-0000001d", "s,a2") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx_builtins.c: Goto (from-trunk,s,2)
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-trunk:2] Answer("SIP/206.189.202.87-0000001d", "") in new stack
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-trunk:3] Log("SIP/206.189.202.87-0000001d", "WARNING,Friendly Scanner from 47.14.82.99") in new stack
[2024-06-13 16:38:58] WARNING[3532][C-0000003a] Ext. s: Friendly Scanner from 47.14.82.99
[2024-06-13 16:38:58] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-trunk:4] Wait("SIP/206.189.202.87-0000001d", "2") in new stack
[2024-06-13 16:39:00] VERBOSE[3532][C-0000003a] pbx.c: Executing [s@from-trunk:5] Playback("SIP/206.189.202.87-0000001d", "ss-noservice") in new stack
[2024-06-13 16:39:00] VERBOSE[3532][C-0000003a] file.c: <SIP/206.189.202.87-0000001d> Playing 'ss-noservice.g722' (language 'en')

You need to define an incoming route for DID 7743342638.

You’re describing an issue with your outbound calls saying that inbound works fine but the logs indicate an inbound call that does not complete.

A lot of weird stuff here.

Assuming that the call to a VZW mobile is what you attempted, it somehow got treated as anonymous SIP (on chan_sip no less).

What is the Context for the calling extension? It should be from-internal, the default.
Can you successfully call *43 (echo test)? If so, can you call other internal extensions?
What’s with the SIREN codec? That shouldn’t be enabled anywhere.

You are also permitting guest incoming calls, which wouldn’t normally be necessary if you were to use chan_pjsip. I wonder if a local extension is matching against your anonymous caller match rule, rather than a specific one.

I tried calling my wifes cellphone from the microsip connected to freepbx. It didnt work. The *43 gives the same “The number you have dialed is not in service”

Yea I enabled anonymous and disabledthe firewall just to check

I will be having a plumbing business where 100 calls a day from random numbers. How would that work then if every number has to be defined?

At the Asterisk command prompt, type
pjsip set logger on
sip set debug on
Make a call to *43, paste the Asterisk log for the call at pastebin.com and post the link here.

The reason I suggested that the number be defined was because the log showed the call as being treated as incoming. My second reply suggested an alternative theory in which an outgoing call was being matched as incoming.

[2024-06-13 17:50:44] VERBOSE[1751] chan_sip.c: <--- SIP read from UDP:47.14.8 - Pastebin.com PS. Appreciate you helping me along with David !

So ext. 101 is connecting to chan_sip on port 5160. Is that intentional? If so, please post your chan_sip extension settings. Do you have a context setting (other than from-internal) somewhere? If so, why?

If this is intended to be a pjsip extension, you should be connecting to port 5060.

Im totally new to pbx systems and first time setting this all up! The only way I got microsip to connect was adding :5160 at the end of the ip address. I read that on reddit somewhere. I am not sure where my chan_sip extension settings are? I can change my microsip phone back to 5060. When I do it just has “request timeout” after a minute when I try dialing.

By default, your PBX has pjsip listening on port 5060 and chan_sip on 5160. When you created extension 101, which did you specify? If you don’t have an extension 101, I don’t see how incoming is working, but maybe you have some workaround.

In Asterisk SIP settings, pjsip tab, what is Port to Listen On? On chan_sip tab, what is Bind Port?

In that case you should disable chan_sip, as it is obsolete;