[asterisk-users] dnd not working correctly
Ott Rose
sixfourimpala at hotmail.com
Tue Mar 30 08:37:07 CDT 2010
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.
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.
Date: Mon, 29 Mar 2010 23:58:43 -0600
From: alyed at vivoxie.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
I'm not an Amportal expert so all I can say from:
> -- Executing [*76 at from-internal:8] Playback("SIP/117-000001f6",
"do-not-disturb&activated") in new stack
> -- Executing [*76 at from-internal:9] Macro("SIP/117-000001f6",
"hangupcall,") in new stack
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.
Just to be sure play this sound file independently.
Sorry but other than this there's little I can do, maybe someone else has experience with this.
Alyed
2010/3/29 Ott Rose <sixfourimpala at hotmail.com>
i posted this on the freepbx site. here is the response
"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."
so i checked /etc/log/asterisk/full
and in vi full i did /error and /117 (my ext) and /activate didn't really find anything
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?
Date: Fri, 26 Mar 2010 18:34:46 -0600
From: alyed at vivoxie.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
Seems like an Amportal configration problem not and Asterisk issue. Maybe you should try in one of the FreePBX users list.
Alyed
2010/3/26 Ott Rose <sixfourimpala at hotmail.com>
i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Extension Changed 117[ext-local] new state InUse for Notify User 102
== Extension Changed 117[ext-local] new state InUse for Notify User 103
== Extension Changed 117[ext-local] new state InUse for Notify User 114
-- Executing [*76 at from-internal:1] Answer("SIP/117-000001f6", "") in new stack
-- Executing [*76 at from-internal:2] Wait("SIP/117-000001f6", "1") in new stack
-- Executing [*76 at from-internal:3] Macro("SIP/117-000001f6", "user-callerid,") in new stack
-- Executing [s at macro-user-callerid:1] Set("SIP/117-000001f6", "AMPUSER=117") in new stack
-- Executing [s at macro-user-callerid:2] GotoIf("SIP/117-000001f6", "0?report") in new stack
-- Executing [s at macro-user-callerid:3] ExecIf("SIP/117-000001f6", "1?Set(REALCALLERIDNUM=117)") in new stack
-- Executing [s at macro-user-callerid:4] Set("SIP/117-000001f6", "AMPUSER=117") in new stack
-- Executing [s at macro-user-callerid:5] Set("SIP/117-000001f6", "AMPUSERCIDNAME=My Name") in new stack
-- Executing [s at macro-user-callerid:6] GotoIf("SIP/117-000001f6", "0?report") in new stack
-- Executing [s at macro-user-callerid:7] Set("SIP/117-000001f6", "AMPUSERCID=117") in new stack
-- Executing [s at macro-user-callerid:8] Set("SIP/117-000001f6", "CALLERID(all)="My Name" <117>") in new stack
-- Executing [s at macro-user-callerid:9] GotoIf("SIP/117-000001f6", "0?continue") in new stack
-- Executing [s at macro-user-callerid:10] Set("SIP/117-000001f6", "__TTL=64") in new stack
-- Executing [s at macro-user-callerid:11] GotoIf("SIP/117-000001f6", "1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s at macro-user-callerid:18] NoOp("SIP/117-000001f6", "Using CallerID "My Name" <117>") in new stack
-- Executing [*76 at from-internal:4] GotoIf("SIP/117-000001f6", "1?activate:deactivate") in new stack
-- Goto (from-internal,*76,5)
-- Executing [*76 at from-internal:5] Set("SIP/117-000001f6", "DB(DND/117)=YES") in new stack
-- Executing [*76 at from-internal:6] Set("SIP/117-000001f6", "STATE=BUSY") in new stack
-- Executing [*76 at from-internal:7] Gosub("SIP/117-000001f6", "app-dnd-toggle,sstate,1") in new stack
-- Executing [sstate at app-dnd-toggle:1] Set("SIP/117-000001f6", "DEVICE_STATE(Custom:DND117)=BUSY") in new stack
-- Executing [sstate at app-dnd-toggle:2] Set("SIP/117-000001f6", "DEVICES=117") in new stack
-- Executing [sstate at app-dnd-toggle:3] GotoIf("SIP/117-000001f6", "0?return") in new stack
== Extension Changed 117[ext-local] new state Busy for Notify User 102
-- Executing [sstate at app-dnd-toggle:4] Set("SIP/117-000001f6", "LOOPCNT=1") in new stack
-- Executing [sstate at app-dnd-toggle:5] Set("SIP/117-000001f6", "ITER=1") in new stack
-- Executing [sstate at app-dnd-toggle:6] Set("SIP/117-000001f6", "DEVICE_STATE(Custom:DEVDND117)=BUSY") in new stack
== Extension Changed 117[ext-local] new state Busy for Notify User 103
== Extension Changed 117[ext-local] new state Busy for Notify User 114
-- Executing [sstate at app-dnd-toggle:7] Set("SIP/117-000001f6", "ITER=2") in new stack
-- Executing [sstate at app-dnd-toggle:8] GotoIf("SIP/117-000001f6", "0?begin") in new stack
-- Executing [sstate at app-dnd-toggle:9] Return("SIP/117-000001f6", "") in new stack
-- Executing [*76 at from-internal:8] Playback("SIP/117-000001f6", "do-not-disturb&activated") in new stack
-- Executing [*76 at from-internal:9] Macro("SIP/117-000001f6", "hangupcall,") in new stack
-- Executing [s at macro-hangupcall:1] GotoIf("SIP/117-000001f6", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s at macro-hangupcall:4] GotoIf("SIP/117-000001f6", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s at macro-hangupcall:7] GotoIf("SIP/117-000001f6", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] Hangup("SIP/117-000001f6", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/117-000001f6' in macro 'hangupcall'
== Spawn extension (from-internal, *76, 9) exited non-zero on 'SIP/117-000001f6'
-- Executing [h at from-internal:1] Macro("SIP/117-000001f6", "hangupcall") in new stack
-- Executing [s at macro-hangupcall:1] GotoIf("SIP/117-000001f6", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s at macro-hangupcall:4] GotoIf("SIP/117-000001f6", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s at macro-hangupcall:7] GotoIf("SIP/117-000001f6", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] Hangup("SIP/117-000001f6", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/117-000001f6' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/117-000001f6'
phoneserver*CLI>
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.
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