What i need to delete from backup(click trash) ?

I understand that direct media is the cause of the issue as well as the fix and that call recording disabled direct media. That’s why external calls work as well as as calls when recording was enabled. There were no changes made prior to this issue starting.

I am trying to determine two things.

1. Why this started suddenly on Tuesday
2. Why it only only occurs with certain public IP addresses. The issue is not happening on all calls.

It’s v13, the Backup/Restore will go all the way back to 2.10 or something like that. So I guess I’m missing how this isn’t working. v13 can let you take backups, is that part not working?

Did you trunk also have Direct Media enabled? Direct Media only works when both endpoints configured in Asterisk have it enabled. If one has it disabled, it doesn’t work.

No idea, we haven’t seen any real troubleshooting data from the system.

It doesn’t occur with certain IP addresses, it occurs with certain networks. Again, we haven’t seen anything that tells us any real details. But one possible answer to this question is, Phone A calls Phone B over the Internet via the PBX with Direct Media on. Phone A has RFC1918 (private) IPs listed in its Contact or SDP information, possible Phone B does as well. You can’t send media over the public Internet if the destination is 192.168.1.124 (or any private space).

So depending on the devices involved and the network they are on (which will be NAT’d) some of them can be sending private IPs in the SIP packet when there should be public IPs and since there is nothing in between to repair said NAT issues, they become NAT issues at either side.

Turning Direct Media off or using certain features of Asterisk makes Asterisk stay in the media path which then allows Asterisk to fix all the broken NAT your devices use.

We have tried by enabling NAT also, but the issue is still same.

What firmware level is your fortigate on? Based on the level it is on, you will want to disable VOIP ALG as per firmware level as per instructions here or similar:

As per @BlazeStudios studios and my recommendation, make sure you have a virtual IP designated that maps DIRECTLY to your internal IP via a VIP mapping; this is very important and will make the difference. It does not matter if it’s your 1 External Class C identified by your ISP or a class C from a block you own, but its very important you directly VIP MAP the Class C External to your Internal IP and then use that for the incoming fortigate rule from external to internal on your fortigate. More than likely, if you do NOT do this, you will not get traffic to your PBX. make sure as well to disable your FreePBX SOftware firewall since you are using the fortigate as your firewall and ONLY allowing your sip gateway to hit this rule. DO NOT allow ANY other EXTERNAL IPs access to this rule ONLY allow your SIP GATEWAY provider.

As well you say ALL Services, I’ll assume you mean required SIP and RTP services and what whatever your SIP gateway provider is asking for. As an example, here is what Thinktel requires the SIP gateway to have access to on a FreePBX, you will have to check PNG to make sure you have all the required services, ONLY allow the services needed:

One last note, in this section here Under Settings–>Asterisk SIP Settings on FreePBX, make sure you Define your EXTERNAL IP you VIP Mapped, and the internal segments of your network for the VIP Map and NAT IE:

ALSO, make sure your local segments are WHITELISTED under Admin–>Sysadmin–>Intrusion Detection:

This should be enough to make things work; BTW, this part is all in the documentation. The ALG stuff may not be, but there it is.

I tried doing those things, and then rebooting the server for good measure yesterday, but today the second extension is no longer enabled. It seems to happen when they try to log in from the email. Is there some kind of blacklist on the sangoma server that sends the invites?

I did see an update to the sangoma connect module this morning. Now upgraded to 16.0.47.9

Sangoma Realtime API verison 16.0.49.12

I have an issue with my freepbx system. Some incoming calls are fine but for others, it just says cannot complete or number not found. Below are my log, hope anyone can help. thanks.

2651	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: Ended up with real PJSIP Dial string PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079	
2652	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: dbset CALLTRACE/902 to +17788589566	
2653	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: Discovered PJSIP Endpoint PJSIP/903	
2654	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: Ended up with real PJSIP Dial string PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911	
2655	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: dbset CALLTRACE/903 to +17788589566	
2656	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: Discovered PJSIP Endpoint PJSIP/999	
2657	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: Ended up with real PJSIP Dial string PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa	
2658	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: dbset CALLTRACE/999 to +17788589566	
2659	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: Filtered ARG3: 902-903-999	
2660	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: RVOL_MODE ''	
2661	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: RVOL is:	
2662	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: agi://127.0.0.1/dialparties.agi: RVOLPARENT is:	
2663	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_agi.c: <PJSIP/MySIP-00000013>AGI Script agi://127.0.0.1/dialparties.agi completed, returning 0	
2664	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:8] GotoIf("PJSIP/MySIP-00000013", "1?normdial") in new stack	
2665	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx_builtins.c: Goto (macro-dial,s,11)	
2666	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:11] NoOp("PJSIP/MySIP-00000013", "Returned from dialparties with groups to dial") in new stack	
2667	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:12] NoOp("PJSIP/MySIP-00000013", "ringall array ") in new stack	
2668	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:13] NoOp("PJSIP/MySIP-00000013", "ds= PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079&PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911&PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa,120,m(RED-Music)htM(auto-blkvm) ") in new stack	
2669	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:14] NoOp("PJSIP/MySIP-00000013", "dsextra= ") in new stack	
2670	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:15] Set("PJSIP/MySIP-00000013", "ds=PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079&PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911&PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa,120,m(RED-Music)htM(auto-blkvm)") in new stack	
2671	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:16] NoOp("PJSIP/MySIP-00000013", "ds= PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079&PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911&PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa,120,m(RED-Music)htM(auto-blkvm)") in new stack	
2672	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:17] Set("PJSIP/MySIP-00000013", "__FMGL_DIAL=") in new stack	
2673	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:18] Set("PJSIP/MySIP-00000013", "LOOPCNT=3") in new stack	
2674	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:19] Set("PJSIP/MySIP-00000013", "ITER=1") in new stack	
2675	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:20] Set("PJSIP/MySIP-00000013", "__EXTTOCALL=902") in new stack	
2676	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:21] Set("PJSIP/MySIP-00000013", "__MCEXTTOCALL=902") in new stack	
2677	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:22] NoOp("PJSIP/MySIP-00000013", "Working with 902") in new stack	
2678	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:23] ExecIf("PJSIP/MySIP-00000013", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack	
2679	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:24] ExecIf("PJSIP/MySIP-00000013", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack	
2680	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:25] Set("PJSIP/MySIP-00000013", "ITER=2") in new stack	
2681	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:26] GotoIf("PJSIP/MySIP-00000013", "1?ndloopbegin") in new stack	
2682	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx_builtins.c: Goto (macro-dial,s,20)	
2683	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:20] Set("PJSIP/MySIP-00000013", "__EXTTOCALL=903") in new stack	
2684	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:21] Set("PJSIP/MySIP-00000013", "__MCEXTTOCALL=903") in new stack	
2685	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:22] NoOp("PJSIP/MySIP-00000013", "Working with 903") in new stack	
2686	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:23] ExecIf("PJSIP/MySIP-00000013", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack	
2687	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:24] ExecIf("PJSIP/MySIP-00000013", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack	
2688	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:25] Set("PJSIP/MySIP-00000013", "ITER=3") in new stack	
2689	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:26] GotoIf("PJSIP/MySIP-00000013", "1?ndloopbegin") in new stack	
2690	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx_builtins.c: Goto (macro-dial,s,20)	
2691	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:20] Set("PJSIP/MySIP-00000013", "__EXTTOCALL=999") in new stack	
2692	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:21] Set("PJSIP/MySIP-00000013", "__MCEXTTOCALL=999") in new stack	
2693	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:22] NoOp("PJSIP/MySIP-00000013", "Working with 999") in new stack	
2694	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:23] ExecIf("PJSIP/MySIP-00000013", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack	
2695	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:24] ExecIf("PJSIP/MySIP-00000013", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack	
2696	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:25] Set("PJSIP/MySIP-00000013", "ITER=4") in new stack	
2697	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:26] GotoIf("PJSIP/MySIP-00000013", "0?ndloopbegin") in new stack	
2698	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:27] Macro("PJSIP/MySIP-00000013", "dial-ringall-predial-hook,") in new stack	
2699	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial-ringall-predial-hook:1] MacroExit("PJSIP/MySIP-00000013", "") in new stack	
2700	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:28] ExecIf("PJSIP/MySIP-00000013", "0?Set(CWRING=r(callwaiting)):Set(CWRING=)") in new stack	
2701	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:29] ExecIf("PJSIP/MySIP-00000013", "0?Set(ds=PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079&PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911&PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa,120,m(RED-Music)htM(auto-blkvm)g)") in new stack	
2702	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@macro-dial:30] Dial("PJSIP/MySIP-00000013", "PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079&PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911&PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa,120,m(RED-Music)htM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack	
2703	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/902-00000014 Internal Gosub(func-apply-sipheaders,s,1) start	
2704	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/902-00000014", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack	
2705	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/902-00000014", "Applying SIP Headers to channel PJSIP/902-00000014") in new stack	
2706	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/902-00000014", "localchan=902-00000014") in new stack	
2707	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:4] Set("PJSIP/902-00000014", "DialMCEXT=902") in new stack	
2708	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:5] Set("PJSIP/902-00000014", "CHANNEL(hangup_handler_push)=app-missedcall-hangup,902,1") in new stack	
2709	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:6] Set("PJSIP/902-00000014", "TECH=PJSIP") in new stack	
2710	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:7] Set("PJSIP/902-00000014", "SIPHEADERKEYS=") in new stack	
2711	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:8] While("PJSIP/902-00000014", "0") in new stack	
2712	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_while.c: Jumping to priority 14	
2713	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:15] Return("PJSIP/902-00000014", "") in new stack	
2714	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: Spawn extension (from-internal, 9988, 1) exited non-zero on 'PJSIP/902-00000014'	
2715	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/902-00000014 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=	
2716	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/903-00000015 Internal Gosub(func-apply-sipheaders,s,1) start	
2717	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/903-00000015", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack	
2718	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/903-00000015", "Applying SIP Headers to channel PJSIP/903-00000015") in new stack	
2719	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/903-00000015", "localchan=903-00000015") in new stack	
2720	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:4] Set("PJSIP/903-00000015", "DialMCEXT=903") in new stack	
2721	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:5] Set("PJSIP/903-00000015", "CHANNEL(hangup_handler_push)=app-missedcall-hangup,903,1") in new stack	
2722	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:6] Set("PJSIP/903-00000015", "TECH=PJSIP") in new stack	
2723	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:7] Set("PJSIP/903-00000015", "SIPHEADERKEYS=") in new stack	
2724	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:8] While("PJSIP/903-00000015", "0") in new stack	
2725	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_while.c: Jumping to priority 14	
2726	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:15] Return("PJSIP/903-00000015", "") in new stack	
2727	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: Spawn extension (from-internal, 9988, 1) exited non-zero on 'PJSIP/903-00000015'	
2728	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/903-00000015 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=	
2729	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/999-00000016 Internal Gosub(func-apply-sipheaders,s,1) start	
2730	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/999-00000016", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack	
2731	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/999-00000016", "Applying SIP Headers to channel PJSIP/999-00000016") in new stack	
2732	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/999-00000016", "localchan=999-00000016") in new stack	
2733	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:4] Set("PJSIP/999-00000016", "DialMCEXT=999") in new stack	
2734	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:5] Set("PJSIP/999-00000016", "CHANNEL(hangup_handler_push)=app-missedcall-hangup,999,1") in new stack	
2735	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:6] Set("PJSIP/999-00000016", "TECH=PJSIP") in new stack	
2736	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:7] Set("PJSIP/999-00000016", "SIPHEADERKEYS=") in new stack	
2737	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:8] While("PJSIP/999-00000016", "0") in new stack	
2738	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_while.c: Jumping to priority 14	
2739	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@func-apply-sipheaders:15] Return("PJSIP/999-00000016", "") in new stack	
2740	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: Spawn extension (from-internal, 9988, 1) exited non-zero on 'PJSIP/999-00000016'	
2741	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/999-00000016 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=	
2742	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: Called PJSIP/902/sip:902@50.98.4.244:1024;line=abw6mpsd;x-ast-orig-host=192.168.2.10:3079	
2743	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: Called PJSIP/903/sip:903@50.98.4.244:55911;ob;x-ast-orig-host=192.168.2.5:55911	
2744	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: Called PJSIP/999/sip:999@50.98.4.244:47991;rinstance=40b5ed179837eefa	
2745	[2024-05-24 09:57:47] VERBOSE[19838] netsock2.c: Using SIP RTP Audio TOS bits 184	
2746	[2024-05-24 09:57:47] VERBOSE[19838] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.	
2747	[2024-05-24 09:57:47] VERBOSE[19838] netsock2.c: Using SIP RTP Audio CoS mark 5	
2748	[2024-05-24 09:57:47] VERBOSE[19736] netsock2.c: Using SIP RTP Audio TOS bits 184	
2749	[2024-05-24 09:57:47] VERBOSE[19736] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.	
2750	[2024-05-24 09:57:47] VERBOSE[19736] netsock2.c: Using SIP RTP Audio CoS mark 5	
2751	[2024-05-24 09:57:47] VERBOSE[22163] netsock2.c: Using SIP RTP Audio TOS bits 184	
2752	[2024-05-24 09:57:47] VERBOSE[22163] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.	
2753	[2024-05-24 09:57:47] VERBOSE[22163] netsock2.c: Using SIP RTP Audio CoS mark 5	
2754	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] res_musiconhold.c: Started music on hold, class 'RED-Music', on channel 'PJSIP/MySIP-00000013'	
2755	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: PJSIP/999-00000016 connected line has changed. Saving it until answer for PJSIP/MySIP-00000013	
2756	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: PJSIP/902-00000014 connected line has changed. Saving it until answer for PJSIP/MySIP-00000013	
2757	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: PJSIP/903-00000015 connected line has changed. Saving it until answer for PJSIP/MySIP-00000013	
2758	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: PJSIP/903-00000015 is ringing	
2759	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_dial.c: PJSIP/902-00000014 is ringing	
2760	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/999-00000016 Internal Gosub(app-missedcall-hangup,999,1) start	
2761	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:1] NoOp("PJSIP/999-00000016", "Dialed: 999") in new stack	
2762	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:2] NoOp("PJSIP/999-00000016", "Caller: ") in new stack	
2763	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:3] GotoIf("PJSIP/999-00000016", "0?exit") in new stack	
2764	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:4] Set("PJSIP/999-00000016", "EXTENNUM=999") in new stack	
2765	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:5] Set("PJSIP/999-00000016", "FEXTENNUM=999") in new stack	
2766	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:6] GotoIf("PJSIP/999-00000016", "1?exit") in new stack	
2767	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx_builtins.c: Goto (app-missedcall-hangup,999,8)	
2768	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [999@app-missedcall-hangup:8] Return("PJSIP/999-00000016", "") in new stack	
2769	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: Spawn extension (from-internal, 9988, 1) exited non-zero on 'PJSIP/999-00000016'	
2770	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/999-00000016 Internal Gosub(app-missedcall-hangup,999,1) complete GOSUB_RETVAL=	
2771	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/999-00000016 Internal Gosub(crm-hangup,s,1) start	
2772	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:1] NoOp("PJSIP/999-00000016", "Sending Hangup to CRM") in new stack	
2773	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:2] NoOp("PJSIP/999-00000016", "HANGUP CAUSE: 17") in new stack	
2774	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:3] ExecIf("PJSIP/999-00000016", "0?Set(__CRM_VOICEMAIL=)") in new stack	
2775	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:4] NoOp("PJSIP/999-00000016", "MASTER CHANNEL: 1716569867.22 = 1716569866.19") in new stack	
2776	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:5] GotoIf("PJSIP/999-00000016", "1?return") in new stack	
2777	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx_builtins.c: Goto (crm-hangup,s,8)	
2778	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:8] Return("PJSIP/999-00000016", "") in new stack	
2779	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: Spawn extension (from-internal, 9988, 1) exited non-zero on 'PJSIP/999-00000016'	
2780	[2024-05-24 09:57:47] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/999-00000016 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=	
2781	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] res_musiconhold.c: Stopped music on hold on PJSIP/MySIP-00000013	
2782	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/902-00000014 Internal Gosub(app-missedcall-hangup,902,1) start	
2783	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:1] NoOp("PJSIP/902-00000014", "Dialed: 902") in new stack	
2784	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:2] NoOp("PJSIP/902-00000014", "Caller: ") in new stack	
2785	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:3] GotoIf("PJSIP/902-00000014", "0?exit") in new stack	
2786	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:4] Set("PJSIP/902-00000014", "EXTENNUM=902") in new stack	
2787	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:5] Set("PJSIP/902-00000014", "FEXTENNUM=2") in new stack	
2788	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:6] GotoIf("PJSIP/902-00000014", "0?exit") in new stack	
2789	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:7] AGI("PJSIP/902-00000014", "agi://127.0.0.1/missedcallnotify.php,2,,2,0,,PJSIP/902-00000014,,,9988,") in new stack	
2790	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] res_agi.c: <PJSIP/902-00000014>AGI Script agi://127.0.0.1/missedcallnotify.php completed, returning 0	
2791	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [902@app-missedcall-hangup:8] Return("PJSIP/902-00000014", "") in new stack	
2792	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] app_stack.c: Spawn extension (from-internal, 9988, 1) exited non-zero on 'PJSIP/902-00000014'	
2793	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/902-00000014 Internal Gosub(app-missedcall-hangup,902,1) complete GOSUB_RETVAL=	
2794	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] app_stack.c: PJSIP/902-00000014 Internal Gosub(crm-hangup,s,1) start	
2795	[2024-05-24 09:57:49] VERBOSE[15049][C-00000008] pbx.c: Executing [s@crm-hangup:1] NoOp("PJSIP/902-00000014", "Sending Hangup to CRM") in new stack	

Your logging stops before FreePBX has said how it considers the call to have failed, and isn’t detailed enough to see why the individual outgoing calls failed. All one can tell is that they failed quickly, but after the channel was allocated.

Hi everyone, I’m new around here and I need to know if possible the steps for make the upgrade from a freepbx 12 to version 16, and if all the licenses purchased in the old system remain valid. Thanks in avance

The best way forward would be to install a new instance of 16 on a brand new box, take a full backup of 12 and restore it to 16. Then you can do Zend reset on the Sangoma portal and activate your new installation to the same deployment.

If you can push it off though, 17 is going to be released some time this year and the upgrade from 16 to 17 is going to be basically the same path so might as well wait a few months and move to 17 instead.

Thanks a lot, I’m going to work on it. I’ll keep updated

Just out of Curiosity, are you running a FreePBX17 BETA on Debian 12 box? On FreePBX 14, 15 and 16 I used a small script to keep CentOS and FreePBX up to date (A member here had a good script and it NEVER failed me). I Never had an issue with it, but I just did basic OS updates and then Module updates. My question would be now that we are on Debian 12 with FreePBX17 BETA, is it, “Safe” for me to do Standard Debian Updates, and as long as I am NOT upgrading to a new Debian version, is there any fear of Debian 12 getting too far ahead of FreePBX 17, or as long as I do FreePBX module updates alongside my Current Version OS updates, should I be good? My goal in testing the beta is to keep the system and modules as update to date as possible, to see if anything breaks and how often.

So instead of yum you are using apt.

apt update && apt upgrade -y

The FreePBX module upgrade process remains the same

fwconsole ma updateall

Yes, that is exactly what I am doing ha ha ha. Perfect.

You can’t just ‘enable NAT’ you have to also configure’ it (likely with iptables (read firewall) rules.) Mostly rules for SIP (VOIP) REGISTER and INVITE and other for any SDP (AUDIO) connections, SIP might be ‘transported’ on UDP, TCP, TLS. WS, WSS , maybe more , SDP (audio) in asterisk is usually apparent on a mathematically incorrect range of 10000-20000)

Such rules allow for calls to arrive on your ‘external IP’ and be handled by your PBX, any resultant audio/video connections and reverse connections are mostly accepted and properly routed then.

Can we presume that incoming calls work?

Thanks a lot, I appreciate!

Hi @1mrpeter,
Can you try to see if it works correctly for you?

Okay I changed it to the correct IP address. I can call some businesses and it works perfectly however when I try to call personal cellphones there is no audio on either end.

It works perfectly! Thank you!

Took me a moment to realize how to do it. Downloaded the source to /var/www/html/admin/modules/superfecta/sources