Hello,
I would like to hear a busy signal instead of playing a message.
In “Route Congestion Messages” I set Congestion Tones, but they do not work
FreePBX 14
Asterisk 13
Logs - we need logs. /var/log/asterisk/full - show us an example so we can make a recommendation.
Sorry for that.
Extension does not exist
LOG - Extension does not exist
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx.c: Executing [335553202@from-pstn:1] Set(“PJSIP/gw2-0000000c”, “__FROM_DID=335553202”) in new stack
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx.c: Executing [335553202@from-pstn:2] NoOp(“PJSIP/gw2-0000000c”, “Received an unknown call with DID set to 335553202”) in new stack
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx.c: Executing [335553202@from-pstn:3] Goto(“PJSIP/gw2-0000000c”, “s,a2”) in new stack
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx_builtins.c: Goto (from-pstn,s,2)
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@from-pstn:2] Answer(“PJSIP/gw2-0000000c”, “”) in new stack
[2018-08-07 08:28:39] WARNING[19123][C-0000000b] chan_sip.c: This function can only be used on SIP channels.
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@from-pstn:3] Log(“PJSIP/gw2-0000000c”, "WARNING,Friendly Scanner from ") in new stack
[2018-08-07 08:28:39] WARNING[19123][C-0000000b] Ext. s: Friendly Scanner from
[2018-08-07 08:28:39] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@from-pstn:4] Wait(“PJSIP/gw2-0000000c”, “2”) in new stack
[2018-08-07 08:28:41] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@from-pstn:5] Playback(“PJSIP/gw2-0000000c”, “ss-noservice”) in new stack
[2018-08-07 08:28:41] VERBOSE[19123][C-0000000b] file.c: <PJSIP/gw2-0000000c> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Executing [h@from-pstn:1] Macro(“PJSIP/gw2-0000000c”, “hangupcall,”) in new stack
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/gw2-0000000c”, “1?theend”) in new stack
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/gw2-0000000c”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/gw2-0000000c”, " monior file= ") in new stack
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/gw2-0000000c”, “attendedtransfer-rec-restart.php,”) in new stack
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] res_agi.c: <PJSIP/gw2-0000000c>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/gw2-0000000c”, “”) in new stack
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/gw2-0000000c’ in macro ‘hangupcall’
[2018-08-07 08:28:44] VERBOSE[19123][C-0000000b] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘PJSIP/gw2-0000000c’
Also, I would like to hear a busy signal as the internal number is busy, DND is pressed etc.
Is it possible to completely disable voice messages in case of a wrong number, etc.
By default, FreePBX has three error announcements. Assuming English, a number that can’t be routed (nonexistent extension, external call not matching any Outbound Route) gets “Your call cannot be completed as dialed …”, a number rejected as ‘not found’ by the trunking provider gets “The number you have dialed is not in service …” and an error response from the trunking provider (after trying any backup trunks) gets “All circuits are busy now.” In addition, Asterisk sends an error code that will cause most IP phones to display an on-screen message such as “Busy here” or “Service unavailable.”
If desired, you could replace the audio files with ones that play a tone pattern. This might be useful for calls initiated by a robot that could interpret the tones, or in an environment where there is no natural language understood by all users. IMO, the latter case would be better served with bilingual announcements. If you must use tones, choose “reorder tone” or something else, other than a busy signal. Most users understand a busy signal to mean that the called party is on the phone. They will typically retry the same number every few minutes, perhaps for an hour or two, then give up. They may never realize that they were dialing an invalid number.
If an extension is in do-not-disturb, or is busy with call waiting disabled, and voicemail is disabled, Asterisk simply sends a 480 Busy Here status. What gets played or displayed is up to your SIP device. Most IP phones will play a few cycles of a busy signal then automatically hang up.
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