Hello Team,
I have configure SIP Extensions on the FreePBX 15.
I have configured Raspberry Pi with linphone to auto answer the call and thus it acts as a Public announcement system. Also it gets registered to the FreePBX a SIP extension.
Upon dialling the Sip Ext, Raspberry Pi Auto answers the call but no Announcement sound is coming out at the Raspberry Pi end.
I am able to play the mp3 file and hear the sound but not announcement.
Please guide me with what settings do I need to check ?
Assuming the call is connecting (check the logs): https://wiki.freepbx.org/display/SUP/Providing+Great+Debug
Then it is usually network/firewall settings (one way/ no audio). Another thing it could be is a codec issue.
More information is needed to offer meaningful suggestions.
if the Pi is the server, you can omit the softphone and enable the channel_console ALSA driver to autoanswer using the 3.5mm jack as an output,
I love the entire SBC ecosystem. I use Pi’s for a variety of things because they are convenient and significantly cheaper than other options for some tasks.
But in this case, you buy a Pi, card, case, and power supply you are going to be at $50 minimum.
You can currently buy a Snom PA-1 for $115 on Amazon. This gives you everything you are getting out of a Pi for zero troubleshooting. Spending a single hour on setting up pi will blow out any savings.
Here Pi is PA client configured with Linphone to anutoanswer.
In my FreePBX Server, both channel modules not loaded in /etc/asterisk/modules.conf:
noload = chan_alsa.so
noload = chan_oss.so
I am new to the freepbx and struggling to learn the same.
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] pbx.c: Executing [s@macro-dial-one:55] Dial(“SIP/3999-00000001”, “SIP/4001,HhTtrb(func-apply-sipheaders^s^1)”) in new stack
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] netsock2.c: Using SIP RTP TOS bits 184
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] netsock2.c: Using SIP RTP CoS mark 5
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_stack.c: SIP/4001-00000002 Internal Gosub(func-apply-sipheaders,s,1) start
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp(“SIP/4001-00000002”, “Applying SIP Headers to channel SIP/4001-00000002”) in new stack
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:2] Set(“SIP/4001-00000002”, “TECH=SIP”) in new stack
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/4001-00000002”, “SIPHEADERKEYS=”) in new stack
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:4] While(“SIP/4001-00000002”, “0”) in new stack
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_while.c: Jumping to priority 12
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:13] Return(“SIP/4001-00000002”, “”) in new stack
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_stack.c: Spawn extension (from-internal, 4001, 1) exited non-zero on ‘SIP/4001-00000002’
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_stack.c: SIP/4001-00000002 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_dial.c: Called SIP/4001
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_dial.c: SIP/4001-00000002 is ringing
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] app_dial.c: SIP/4001-00000002 answered SIP/3999-00000001
[2020-09-10 15:07:16] VERBOSE[12007][C-00000004] bridge_channel.c: Channel SIP/4001-00000002 joined ‘simple_bridge’ basic-bridge
[2020-09-10 15:07:16] VERBOSE[12006][C-00000004] bridge_channel.c: Channel SIP/3999-00000001 joined ‘simple_bridge’ basic-bridge
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] bridge_channel.c: Channel SIP/3999-00000001 left ‘simple_bridge’ basic-bridge
[2020-09-10 15:07:29] VERBOSE[12007][C-00000004] bridge_channel.c: Channel SIP/4001-00000002 left ‘simple_bridge’ basic-bridge
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] app_macro.c: Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘SIP/3999-00000001’ in macro ‘dial-one’
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] app_macro.c: Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘SIP/3999-00000001’ in macro ‘exten-vm’
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Spawn extension (from-internal, 4001, 3) exited non-zero on ‘SIP/3999-00000001’
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Executing [h@from-internal:1] Macro(“SIP/3999-00000001”, “hangupcall”) in new stack
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/3999-00000001”, “1?theend”) in new stack
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/3999-00000001”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/3999-00000001”, "SIP/4001-00000002 montior file= ") in new stack
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“SIP/3999-00000001”, “1?skipagi”) in new stack
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“SIP/3999-00000001”, “”) in new stack
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/3999-00000001’ in macro ‘hangupcall’
[2020-09-10 15:07:29] VERBOSE[12006][C-00000004] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/3999-00000001’
Above are the logs when I dial the Sip ext “4001” from ext 3999
Please suggest how to fix the issue os no sound at the receiver end when call is answered
You are right but we can not procure it currently. will think of it in future projects.
Chan console would need to be loaded which will need a working /etc/asterisk/console.conf file (examples on the web or
/usr/src/asterisk*/configs/samples/console.conf.sample
if you compiled yourself. )
The server would of course need to be physically at the location with an ‘audio out’
You can easily install asterisk on the Pi , create a trunk to and then register with your server to use the pi hardware without adding anything but time, (all while learning to fish)
https://sourceforge.net/p/raspbx/discussion/general/thread/6d319f48/
Let me explain the scenario again:
I have Installed and Configured RasPBX on Raspberry Pi 4. Configured Chan_sip extensions.
Installed and configured another Raspberry Pi 3 with Linphone as PA Client which is registered to the RasPBX with the configured Extension.
When I dial the registered ext of PA client, it auto answers the call as configured but the announcement sound is not coming from the headphones connected to the PA Client.
The same set up I have for FreePBX 13 it works perfectly as expected but with FreePBX 15, I am not getting Audio specially announcement upon when PA client auto answers the call.
Able play the MP3 audio file on the PA client.
I have fixed the issue by rectifying NAT settings of FreePBX.
Thank you for your support.