Darn. I guess we’ll see what @lgaetz says about the Debian Version. No big deal either way, I can just re-install the stable version of FreePBX 17 on Debian when it is released if I need to.
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Freepbx 17 problems feedback
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No audio on some PJSIP endpoints
I am using Asterisk 18.9 and have for a long time. This started suddenly on Tuesday without any changes being done. It seems it seems like the "don’t do direct behind NAT stopped working and needs to be explicitly enabled. The weird thing is that the is doesn’t happen with every connection. It seems ISP related.
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Incoming calls not reaching to PBX
Version of FreePBX? You have a NAT set up on your Firewall with a Virtual IP Mapping from External to Internal? Enterprise Firewall model? What services are you allowing thorough the NAT to your FreePBX? Who is the SIP gateway provider? Have you made sure to allow ALL services needed for SIP through the NAT to your FreePBX server? Depending on your firewall model and SIP gateway provider, you may need to Disable VOIP ALG on your firewall, there will be documentation from your firewall manufacturer on how to disable built in ALG Services.
I can provide other suggestions as well, but we need to start somewhere.
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Incoming calls not reaching to PBX
Thank you for your response. Here are the details you requested:
Version of FreePBX: FreePBX 15.0.37.4
You have a NAT set up on your Firewall with a Virtual IP Mapping from External to Internal?: No we havn’t set up NAT on firewall
Enterprise Firewall model : Fortigate-80F
What services are you allowing thorough the NAT to your FreePBX: All Services allowed
Who is the SIP gateway provider: PNG
Have you made sure to allow ALL services needed for SIP through the NAT to your FreePBX server: Yes
VOIP ALG: We have to check our firewall documentation and will disable the built-in VOIP ALG as per the manufacturer’s guidelines.
Please let me know if you need any additional information or if there are other suggestions you have for troubleshooting this issue.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
have you tried this?
From 4-5 years ago i have used. Maybe could be fix your data transfer from Old to new installed PBX.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
That script is dead and no longer supported. It shouldn’t be used for this.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
Thanks, Tom. It was good script.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
Yes but it did what Backup/Restore does as of v15 so not really needed.
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Incoming calls not reaching to PBX
That would be your issue.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
And what i can used for transfer all settings from pbx13 to 16 ?
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No audio on some PJSIP endpoints
Part of the problem here is that for chan_pjsip the extension configuration defaults to having Direct Media enabled in FreePBX. That would mean any of your internal calls would have both endpoints trying to use Direct Media with each other and yes that can be impacted by the Internet connection they are using and other networking factors involved.
Direct Media is automatically disabled if the following are used on the call, Confbidge, MixMonitor (Monitor too for legacy), ChanSpy and various others found here Asterisk PJSIP Troubleshooting Guide - Asterisk Documentation
So when you enabled call recording you immediately forced Direct Media out of the call and that is why there is audio on those calls.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
The same thing you were told earlier in this thread. You create a FreePBX v16 box, you do a backup of the FreePBX v13 box, you do a Restore of the v13 backup on the v16 system.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
@BlazeStudios Tom his PBX is old thats why we are looking some how to send backup from OLD PBX to New Installed PBX-16
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No audio on some PJSIP endpoints
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.
- Why this started suddenly on Tuesday
- Why it only only occurs with certain public IP addresses. The issue is not happening on all calls.
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Correct update FreePBX 13.0.194.10 to FreePBX 16
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?
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No audio on some PJSIP endpoints
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.
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Incoming calls not reaching to PBX
We have tried by enabling NAT also, but the issue is still same.
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Incoming calls not reaching to PBX
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.
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Sangoma Talk disabling extension automatically
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
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SIP trunk inbound calls not working for some numbers
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
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