Broadcast module needs a pause in order to answer calls correctly.

A user is complaining that after the broadcast module launches a call and the call is answered by the PBX , the call is never sent to the desired destination. It remains in silence for 30 seconds and then hangs.

I searched through our Jira tickets and found this one: https://issues.freepbx.org/browse/FREEPBX-15897
This Jira describes a similar behavior the customer is having.

Si looking at the logs, the calls that are failing show the following:

1. [2018-09-26 10:40:00] VERBOSE[12510][C-00003a8d] app_amd.c: AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] maximumWordLength [5000]
2. [2018-09-26 10:40:00] VERBOSE[12618][C-00003a8c] bridge_channel.c: Channel SIP/40015819-00000037 joined 'simple_bridge' basic-bridge <d918089e-b82d-4e30-b124-4fbeb80cf10a>
3. [2018-09-26 10:40:00] VERBOSE[12511][C-00003a8c] bridge_channel.c: Channel Local/987064255@from-internal-00001d48;2 joined 'simple_bridge' basic-bridge <d918089e-b82d-4e30-b124-4fbeb80cf10a>
4. [2018-09-26 10:40:31] NOTICE[2349] chan_sip.c: Disconnecting call 'SIP/40015819-00000037' for lack of RTP activity in 31 seconds
5. [2018-09-26 10:40:31] VERBOSE[12618][C-00003a8c] bridge_channel.c: Channel SIP/40015819-00000037 left 'simple_bridge' basic-bridge <d918089e-b82d-4e30-b124-4fbeb80cf10a>

Please notice the message
[2018-09-26 10:40:31] NOTICE[2349] chan_sip.c: Disconnecting call 'SIP/40015819-00000037' for lack of RTP activity in 31 seconds

This call’s log is here:
https://pastebin.freepbx.org/view/51caa905#L682

This is the original part of the dialplan in Asterisk set by the broadcast module: https://pastebin.freepbx.org/view/f1d2dd89

In the mentioned Jira, Lorne suggests adding the following entry to the dialplan:

exten => s,n,Playback(silence/50ms)

This can be seen here: https://pastebin.freepbx.org/view/87ea337b#L9

After reloading the dialplan, the call is answered successfully with two-way audio and it is transferred to the right destination:

1. [2018-09-25 15:22:06] VERBOSE[2121][C-00000737] app_dial.c: SIP/40015819-00000032 is making progress passing it to Local/987456060@from-internal-0000039b;2
2. [2018-09-25 15:22:06] VERBOSE[2120][C-00000737] app_dial.c: Local/987456060@from-internal-0000039b;1 is making progress passing it to Local/87456060@webcallback-1-0000039a;2
3. [2018-09-25 15:22:15] VERBOSE[2121][C-00000737] app_dial.c: SIP/40015819-00000032 answered Local/987456060@from-internal-0000039b;2
4. [2018-09-25 15:22:15] VERBOSE[2120][C-00000737] app_dial.c: Local/987456060@from-internal-0000039b;1 answered Local/87456060@webcallback-1-0000039a;2
5. [2018-09-25 15:22:15] VERBOSE[2121][C-00000737] bridge_channel.c: Channel Local/987456060@from-internal-0000039b;2 joined 'simple_bridge' basic-bridge <adc7af60-62e2-4ce2-9680-4e0263ddf4c9>
6. [2018-09-25 15:22:15] VERBOSE[2395][C-00000737] bridge_channel.c: Channel Local/987456060@from-internal-0000039b;1 joined 'simple_bridge' basic-bridge <90499821-6187-4a18-b937-8fd765bab1b7>
7. [2018-09-25 15:22:15] VERBOSE[2395][C-00000737] bridge_channel.c: Channel Local/987456060@from-internal-0000039b;2 left 'simple_bridge' basic-bridge <adc7af60-62e2-4ce2-9680-4e0263ddf4c9>
8. [2018-09-25 15:22:15] VERBOSE[2395][C-00000737] bridge_channel.c: Channel Local/87456060@webcallback-1-0000039a;2 swapped with Local/987456060@from-internal-0000039b;2 into 'simple_bridge' basic-bridge <adc7af60-62e2-4ce2-9680-4e0263ddf4c9>

This is the log for the above call:

https://pastebin.freepbx.org/view/93a7d0f8