boop sound when the customer answer the phone

[wwwanaya]

The problem is when the customer answer the call, asterisk or vicidial execute a sound that sound like a "boop"...
The "boop" sound is reproduce also when the agents logging in vicidial, in that moment, the asterisk console show "conf-only person.gsm"... and after that greeting sounds that "boop"...

I want to remove that sound... for the customers.

Here is the console from asterisk:

[Dec 11 11:44:28] -- Executing [144972719486@default:1] NoOp("Local/144972719486@default-00000011;2", "quita prefijo 1") in new stack
[Dec 11 11:44:28] -- Executing [144972719486@default:2] Set("Local/144972719486@default-00000011;2", "nuevaext=44972719486") in new stack
[Dec 11 11:44:28] -- Executing [144972719486@default:3] Goto("Local/144972719486@default-00000011;2", "outbound,44972719486,1") in new stack
[Dec 11 11:44:28] -- Goto (outbound,44972719486,1)
[Dec 11 11:44:28] -- Executing [44972719486@outbound:1] NoOp("Local/144972719486@default-00000011;2", "") in new stack
[Dec 11 11:44:28] -- Executing [44972719486@outbound:2] Dial("Local/144972719486@default-00000011;2", "SIP/972719486@outbound") in new stack
[Dec 11 11:44:28] == Using SIP RTP CoS mark 5
[Dec 11 11:44:28] -- Called SIP/972719486@outbound
[Dec 11 11:44:29] -- SIP/outbound-0000000b is making progress passing it to Local/144972719486@default-00000011;2
[Dec 11 11:44:32] -- SIP/outbound-0000000b is ringing
[Dec 11 11:44:32] -- SIP/outbound-0000000b is making progress passing it to Local/144972719486@default-00000011;2
[Dec 11 11:44:41] -- SIP/outbound-0000000b answered Local/144972719486@default-00000011;2
[Dec 11 11:44:41] > Channel Local/144972719486@default-00000011;1 was answered.
[Dec 11 11:44:41] -- Executing [8368@default:1] Playback("Local/144972719486@default-00000011;1", "sip-silence") in new stack
[Dec 11 11:44:41] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 11 11:44:41] -- <Local/144972719486@default-00000011;1> Playing 'sip-silence.gsm' (language 'en')
[Dec 11 11:44:41] == Spawn extension (outbound, 44972719486, 2) exited non-zero on 'Local/144972719486@default-00000011;2'
[Dec 11 11:44:44] -- Executing [8368@default:2] AGI("SIP/outbound-0000000b", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 11 11:44:44] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=777))
[Dec 11 11:44:44] -- <SIP/outbound-0000000b>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 11 11:44:44] -- Executing [8368@default:3] AGI("SIP/outbound-0000000b", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Dec 11 11:44:44] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 11 11:44:44] -- <SIP/outbound-0000000b>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 11 11:44:44] -- Executing [172*016*049*100*8600052@default:1] Goto("SIP/outbound-0000000b", "default,8600052,1") in new stack
[Dec 11 11:44:44] -- Goto (default,8600052,1)
[Dec 11 11:44:44] -- Executing [8600052@default:1] MeetMe("SIP/outbound-0000000b", "8600052,F") in new stack
[Dec 11 11:44:46] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 11 11:44:46] -- Executing [58600052@default:1] MeetMe("Local/58600052@default-00000012;2", "8600052,Fmq") in new stack
[Dec 11 11:44:46] > Channel Local/58600052@default-00000012;1 was answered.
[Dec 11 11:44:46] -- Executing [8309@default:1] Answer("Local/58600052@default-00000012;1", "") in new stack
[Dec 11 11:44:46] -- Executing [8309@default:2] Monitor("Local/58600052@default-00000012;1", "wav,20141211-104445_44972719486") in new stack
[Dec 11 11:44:46] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 11 11:44:46] -- Executing [8309@default:3] Wait("Local/58600052@default-00000012;1", "3600") in new stack
[Dec 11 11:44:49] == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/outbound-0000000b'
[Dec 11 11:44:49] -- Executing [h@default:1] AGI("SIP/outbound-0000000b", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Dec 11 11:44:49] -- <SIP/outbound-0000000b>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Also tried with replacing "sip-silence.gsm" with a audio but even not execute my sound, and also I try to rename it for Asterisk not find it and do not run. But nothing happens...
Ideas?

Thanks.

[mflorell]

That sound is compiled into the code of Asterisk, so it cannot be easily changed at all. If you want to remove all enter and leave sounds then your only option is to add a "q" to the Meetme entries in the extensions.conf file and reload the dialplan or restart Asterisk.

[wwwanaya]

Thank you so much.
I did exacly that you said... And it works!

Thanks once again.