yes, that's the default location unless there's any change in /etc/asterisk/asterisk.conf<br><br>I think it should be there, cause when it is not asterisk complains with a message letting you know it wasn't able to find it.<br>
<br>>I dont have a working server to look at so i didn't know if i was even looking in the right place.<br>Pls look for them in the server you are actually having the problems with cause I can't remember that sound file being on the official's asterisk release.<br>
<br>Alyed<br><br> <br><div class="gmail_quote">2010/3/30 Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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where are those sound files kept? i looked last night in /var/lib/asterisk/sounds and i didn't see anything named do-not-disturb.<br><br>if its supposed to be in there then thats a problem. I dont have a working server to look at so i didn't know if i was even looking in the right place.<br>
<br><hr>Date: Mon, 29 Mar 2010 23:58:43 -0600<br>From: <a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] dnd not working correctly<br><br>I'm not an Amportal expert so all I can say from:<br><br>> -- Executing [*76@from-internal:8] Playback("SIP/117-000001f6",
"do-not-disturb&activated") in new stack<br>> -- Executing [*76@from-internal:9] Macro("SIP/117-000001f6",
"hangupcall,") in new stack<br><br>is that Asterisk is playing the "do-not-disturb&activated" file (apparently without errors) and then the next instruction is to hangup the call, hence Asterisk hangs it up.<br>
<br>Just to be sure play this sound file independently.<br><br>Sorry but other than this there's little I can do, maybe someone else has experience with this.<br><br>Alyed<br><br><br><div>2010/3/29 Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com" target="_blank">sixfourimpala@hotmail.com</a>></span><br>
<blockquote style="padding-left: 1ex;">
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<br>i posted this on the freepbx site. here is the response <br><br><br>"from the trace, everything is working. Check your asterisk log for file
errors playing back the audio, could be your sound files are not
installed or messed up."<br><br><br><br>so i checked /etc/log/asterisk/full <br><br>and in vi full i did /error and /117 (my ext) and /activate didn't really find anything<br><br>i didn't see anything but i might be over looking it. I did grep error full and it returned some errors but not related to dnd as far as i can tell. is there some place else to look, a better way to search that file, or can someone tell me what i am looking for?<div>
<div></div><div><br><br><br><br><hr>Date: Fri, 26 Mar 2010 18:34:46 -0600<br>From: <a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] dnd not working correctly<br><br>Seems like an Amportal configration problem not and Asterisk issue. Maybe you should try in one of the FreePBX users list.<br><br>Alyed<br><br><br><br><div>2010/3/26 Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com" target="_blank">sixfourimpala@hotmail.com</a>></span><br>
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i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info <br><br>
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)<br>
Verbosity is at least 4<br>
== Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
== Using SIP VRTP TOS bits 136<br>
== Using SIP VRTP CoS mark 6<br>
== Extension Changed 117[ext-local] new state InUse for Notify User 102<br>
== Extension Changed 117[ext-local] new state InUse for Notify User 103<br>
== Extension Changed 117[ext-local] new state InUse for Notify User 114<br>
-- Executing [*76@from-internal:1] Answer("SIP/117-000001f6", "") in new stack<br>
-- Executing [*76@from-internal:2] Wait("SIP/117-000001f6", "1") in new stack<br>
-- Executing [*76@from-internal:3] Macro("SIP/117-000001f6", "user-callerid,") in new stack<br>
-- Executing [s@macro-user-callerid:1] Set("SIP/117-000001f6", "AMPUSER=117") in new stack<br>
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/117-000001f6", "0?report") in new stack<br>
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/117-000001f6", "1?Set(REALCALLERIDNUM=117)") in new stack<br>
-- Executing [s@macro-user-callerid:4] Set("SIP/117-000001f6", "AMPUSER=117") in new stack<br>
-- Executing [s@macro-user-callerid:5] Set("SIP/117-000001f6", "AMPUSERCIDNAME=My Name") in new stack<br>
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/117-000001f6", "0?report") in new stack<br>
-- Executing [s@macro-user-callerid:7] Set("SIP/117-000001f6", "AMPUSERCID=117") in new stack<br>
-- Executing [s@macro-user-callerid:8] Set("SIP/117-000001f6", "CALLERID(all)="My Name" <117>") in new stack<br>
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/117-000001f6", "0?continue") in new stack<br>
-- Executing [s@macro-user-callerid:10] Set("SIP/117-000001f6", "__TTL=64") in new stack<br>
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/117-000001f6", "1?continue") in new stack<br>
-- Goto (macro-user-callerid,s,18)<br>
-- Executing [s@macro-user-callerid:18] NoOp("SIP/117-000001f6", "Using CallerID "My Name" <117>") in new stack<br>
-- Executing [*76@from-internal:4] GotoIf("SIP/117-000001f6", "1?activate:deactivate") in new stack<br>
-- Goto (from-internal,*76,5)<br>
-- Executing [*76@from-internal:5] Set("SIP/117-000001f6", "DB(DND/117)=YES") in new stack<br>
-- Executing [*76@from-internal:6] Set("SIP/117-000001f6", "STATE=BUSY") in new stack<br>
-- Executing [*76@from-internal:7] Gosub("SIP/117-000001f6", "app-dnd-toggle,sstate,1") in new stack<br>
-- Executing [sstate@app-dnd-toggle:1] Set("SIP/117-000001f6", "DEVICE_STATE(Custom:DND117)=BUSY") in new stack<br>
-- Executing [sstate@app-dnd-toggle:2] Set("SIP/117-000001f6", "DEVICES=117") in new stack<br>
-- Executing [sstate@app-dnd-toggle:3] GotoIf("SIP/117-000001f6", "0?return") in new stack<br>
== Extension Changed 117[ext-local] new state Busy for Notify User 102<br>
-- Executing [sstate@app-dnd-toggle:4] Set("SIP/117-000001f6", "LOOPCNT=1") in new stack<br>
-- Executing [sstate@app-dnd-toggle:5] Set("SIP/117-000001f6", "ITER=1") in new stack<br>
-- Executing [sstate@app-dnd-toggle:6] Set("SIP/117-000001f6", "DEVICE_STATE(Custom:DEVDND117)=BUSY") in new stack<br>
== Extension Changed 117[ext-local] new state Busy for Notify User 103<br>
== Extension Changed 117[ext-local] new state Busy for Notify User 114<br>
-- Executing [sstate@app-dnd-toggle:7] Set("SIP/117-000001f6", "ITER=2") in new stack<br>
-- Executing [sstate@app-dnd-toggle:8] GotoIf("SIP/117-000001f6", "0?begin") in new stack<br>
-- Executing [sstate@app-dnd-toggle:9] Return("SIP/117-000001f6", "") in new stack<br>
-- Executing [*76@from-internal:8] Playback("SIP/117-000001f6", "do-not-disturb&activated") in new stack<br>
-- Executing [*76@from-internal:9] Macro("SIP/117-000001f6", "hangupcall,") in new stack<br>
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/117-000001f6", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br>
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/117-000001f6", "1?skipblkvm") in new stack<br>
-- Goto (macro-hangupcall,s,7)<br>
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/117-000001f6", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br>
-- Executing [s@macro-hangupcall:9] Hangup("SIP/117-000001f6", "") in new stack<br>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/117-000001f6' in macro 'hangupcall'<br>
== Spawn extension (from-internal, *76, 9) exited non-zero on 'SIP/117-000001f6'<br>
-- Executing [h@from-internal:1] Macro("SIP/117-000001f6", "hangupcall") in new stack<br>
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/117-000001f6", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br>
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/117-000001f6", "1?skipblkvm") in new stack<br>
-- Goto (macro-hangupcall,s,7)<br>
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/117-000001f6", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br>
-- Executing [s@macro-hangupcall:9] Hangup("SIP/117-000001f6", "") in new stack<br>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/117-000001f6' in macro 'hangupcall'<br>
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/117-000001f6'<br>
phoneserver*CLI><br>
<br><br>when i dial *76 the phone hangs up after one sec. i do not hear dnd activated or anything. The light on the phone doesn't come on and the screen doesn't say dnd. I have Aastra 57i.<br><div><br>
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