IVR + Announcement can't hear audio

Hi All,

I’m just in the process of setting up a new FreePBX 13 box installed on a MS HyperV box from the FreePBX-64bit-10.13.66 iso. It’s to replace an old install that we have been using for a few years. I’ve got my trunk, incoming and outgoing routes, extensions, ring groups and voicemail setup fine and all working, however…

I have been trying to get some announcements up and running and seem to be struggling.

I recorded them using admin/system recordings, record over extension.
The recordings play back fine from the system recording page
Enabling the feature code allows me to listen to the recording by dialing the feature code from an extension.
I have created an announcement from the applications menu and picked the recording I made
I have also set the destination after playback to be my main ring group.

I then change my inbound routes from ring group to announcement.
The call is answered, no sound is played the call is eventually transfered to the ring group but picking up the receiver does not answer the call.

The following from the logs seems to indicate that the audio is being played and when the playback finishes it forwards on to ring group 600 (which is correct)

[2016-06-10 20:15:41] VERBOSE[47743][C-0000006c] pbx_builtins.c: Goto (app-announcement-1,s,1)
[2016-06-10 20:15:41] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:1] GotoIf("SIP/orbtalk_in-00000041", "0?begin") in new stack
[2016-06-10 20:15:41] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:2] Answer("SIP/orbtalk_in-00000041", "") in new stack
[2016-06-10 20:15:42] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:3] Wait("SIP/orbtalk_in-00000041", "1") in new stack
[2016-06-10 20:15:43] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:4] NoOp("SIP/orbtalk_in-00000041", "Playing announcement Daven Not Here") in new stack
[2016-06-10 20:15:43] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:5] Playback("SIP/orbtalk_in-00000041", "custom/davien_not_here,noanswer") in new stack
[2016-06-10 20:15:43] VERBOSE[47743][C-0000006c] file.c: <SIP/orbtalk_in-00000041> Playing 'custom/davien_not_here.slin' (language 'en')
[2016-06-10 20:15:57] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:6] Goto("SIP/orbtalk_in-00000041", "ext-group,600,1") in new stack
[2016-06-10 20:15:57] VERBOSE[47743][C-0000006c] pbx_builtins.c: Goto (ext-group,600,1)

Then after some more normal looking ring group entries I get the following;

[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] app_dial.c: PJSIP/127-00000130 answered SIP/orbtalk_in-00000041
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:1] Set("PJSIP/127-00000130", "__MACRO_RESULT=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:2] Set("PJSIP/127-00000130", "CFIGNORE=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:3] Set("PJSIP/127-00000130", "MASTER_CHANNEL(CFIGNORE)=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:4] Set("PJSIP/127-00000130", "FORWARD_CONTEXT=from-internal") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:5] Set("PJSIP/127-00000130", "MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:6] Macro("PJSIP/127-00000130", "blkvm-clr,") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-blkvm-clr:1] Set("PJSIP/127-00000130", "SHARED(BLKVM,SIP/orbtalk_in-00000041)=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-blkvm-clr:2] Set("PJSIP/127-00000130", "GOSUB_RETVAL=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/127-00000130", "") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:7] ExecIf("PJSIP/127-00000130", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=127/sip:[email protected]:5060)") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:8] ExecIf("PJSIP/127-00000130", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack
[2016-06-10 20:16:02] VERBOSE[47822][C-0000006c] bridge_channel.c: Channel PJSIP/127-00000130 joined 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] bridge_channel.c: Channel SIP/orbtalk_in-00000041 joined 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:02] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:03] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:03] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:04] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:04] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:05] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:05] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:06] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:06] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:07] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:07] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:08] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:08] VERBOSE[47822][C-0000006c] bridge_channel.c: Channel PJSIP/127-00000130 left 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] bridge_channel.c: Channel SIP/orbtalk_in-00000041 left 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] app_macro.c: Spawn extension (macro-dial, s, 17) exited non-zero on 'SIP/orbtalk_in-00000041' in macro 'dial'
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Spawn extension (ext-group, 600, 14) exited non-zero on 'SIP/orbtalk_in-00000041'
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [h@ext-group:1] Macro("SIP/orbtalk_in-00000041", "hangupcall,") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/orbtalk_in-00000041", "1?theend") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/orbtalk_in-00000041", "0?Set(CDR(recordingfile)=)") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/orbtalk_in-00000041", "") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/orbtalk_in-00000041' in macro 'hangupcall'
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on 'SIP/orbtalk_in-00000041'

I am behind a NAT router with 8 static address, I have port 5060 and 10000 - 20000 forwarding in on one of those address and have all outbound traffic from my Freepbx box going through the same IP.

I have tried setting up an IVR instead of the announcement and get the same issue.

Voicemail seems to be working fine. I can hear the recording and record a message with no problem.

Any help, pointers or additional reading suggestions for getting Announcements / IVR working would be greatly appreciated.

Thanks in advance

I am guessing that you are sip fom your sip trunk and pjsip for your extensions.
If yes then go to sip options and enable nat and set the externip or externhost that your box should use also set the localnet parameter.