[asterisk-users] dnd not working correctly

Tue Mar 30 00:58:43 CDT 2010

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.
>
>
> ------------------------------
> Hotmail is redefining busy with tools for the New Busy. Get more from your
> inbox. Sign up now.<http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_2>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
> Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up
> now. <http://clk.atdmt.com/GBL/go/210850552/direct/01/>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100329/d2cff7e8/attachment.htm 