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Voicemail Disable The person at extension

Offline Graham

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Voicemail Disable The person at extension
« on: January 05, 2009, 01:24:02 AM »
I would like to disable the message The person at extension ... 1234 ... is unavailable, it shows you how to do it here but I can't work it out.

http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail

Offline jester

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Re: Voicemail Disable The person at extension
« Reply #1 on: January 05, 2009, 03:46:55 PM »
If you are using SAIL i would look here: http://www.sarkpbx.com and read the DocsWiki

Offline Graham

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Re: Voicemail Disable The person at extension
« Reply #2 on: January 05, 2009, 03:52:53 PM »
I have looked at the docs and can't find anything that talks about disabling that prompt

Offline jester

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Re: Voicemail Disable The person at extension
« Reply #3 on: January 05, 2009, 04:29:42 PM »
Why would you want to totally disable it?! I believe you can set your own busy message with *51* in stead of the standard one... and you could just record a second of silence if you wanted to... but i don't really see the point.

Offline Graham

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Re: Voicemail Disable The person at extension
« Reply #4 on: January 05, 2009, 04:36:51 PM »
This is what we have,

Closed Inbound Route > IVR Menu, Press 1,2,3,4 etc, the same menu we use when open.
User makes selection and they go to another IVR Menu which just says Sorry we closed leave a message after the tone, timeout is set to foward to VoiceMail, which then plays the default The person at extension etc, etc.

What’s the points of recording a second of silence when it can be disabled.

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Re: Voicemail Disable The person at extension
« Reply #5 on: January 05, 2009, 04:53:06 PM »
Provided you have a relatively recent release of SAIL then in your IVR you can send the outcome to *extn.  This will just give a short message followed by the beep.

S

Offline Graham

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Re: Voicemail Disable The person at extension
« Reply #6 on: January 05, 2009, 05:52:01 PM »
I've setup the IVR to points to an extn no issue it goes through to the voicemail and I'll just have to record a message for each one or change the global one.

One issue I still have is with Automation, I have a cluster called ClusterA, TrunkA belongs to ClusterA.

The Closed Time/Date segments belongs to ClusterA and is set for close at 12PM open at 9AM - week, day and month are set to *.

Problem, Closed Inbound Route is set to hangup but it’s still getting passed through to the Open Inbound Route.

Offline SARK devs

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Re: Voicemail Disable The person at extension
« Reply #7 on: January 05, 2009, 06:02:43 PM »
Hi Graham

I'm not sure I explained very well;  but *+extnum will go to the extension vmail amd simply say "please leave your message after the tone".  It doesn't play "The person at yada yada yada..".

Re Clusters...

Can you run a call when it is closed and capture a console log for us?  Do asterisk -rvvvv and then do agi debug.

When you've got the trace, turn agi debugging off with agi no debug.

Thanks

S




Offline Graham

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Re: Voicemail Disable The person at extension
« Reply #8 on: January 05, 2009, 06:12:58 PM »
Asterisk 1.4.21.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.21.1 currently running on server-1 (pid = 8312)
Verbosity was 0 and is now 4
server-1*CLI> agi debug
AGI Debugging Enabled
    -- Executing [4716596@mainmenu:1] AGI("SIP/4717753-09d6f9b0", "selintra|Inbound|4716596") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/4717753-09d6f9b0
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1231175380.12
AGI Tx >> agi_callerid: xxxxxxxxxxxxx
AGI Tx >> agi_calleridname: xxxxxxxxxxxxx
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 4716596
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: mainmenu
AGI Tx >> agi_extension: 4716596
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE RINGDELAY
AGI Tx >> 200 result=1 (0)
AGI Rx << GET VARIABLE FAXDETECT
AGI Tx >> 200 result=1 (2)
AGI Rx << GET VARIABLE CALLRECORD2
AGI Tx >> 200 result=1 (None)
AGI Rx << GET VARIABLE LTERM
AGI Tx >> 200 result=1 (NO)
AGI Rx << SET VARIABLE MOH ""
AGI Tx >> 200 result=1
AGI Rx << SET CALLERID xxxxxxxxxxxxx
AGI Tx >> 200 result=1
AGI Rx << DATABASE GET "STAT" "OCSTAT"
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE REMOTENUM "4716596"
AGI Tx >> 200 result=1
AGI Rx << GET VARIABLE MTIME
AGI Tx >> 200 result=1 (ON)
AGI Rx << GET VARIABLE VOICEINSTR
AGI Tx >> 200 result=1 (NO)
AGI Rx << SET VARIABLE OPEN "YES"
AGI Tx >> 200 result=1
AGI Rx << DATABASE GET "STAT" "IVRSTAT"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE SYSOP
AGI Tx >> 200 result=1 ()
AGI Rx << EXEC Background silence/1
    -- AGI Script Executing Application: (Background) Options: (silence/1)
    -- <SIP/4717753-09d6f9b0> Playing 'silence/1' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << EXEC Wait 1
    -- AGI Script Executing Application: (Wait) Options: (1)
AGI Tx >> 200 result=0
AGI Rx << STREAM FILE usergreeting2001 "12" 0
    -- Playing 'usergreeting2001' (escape_digits=12) (sample_offset 0)
AGI Tx >> 200 result=49 endpos=108320
AGI Rx << GET VARIABLE SYSOP
AGI Tx >> 200 result=1 ()
AGI Rx << SET CALLERID AOE<xxxxxxxxxxxxx>
AGI Tx >> 200 result=1
AGI Rx << EXEC Queue SWIMSTORE-AOE|tThH
    -- AGI Script Executing Application: (Queue) Options: (SWIMSTORE-AOE|tThH)
    -- Started music on hold, class 'default', on SIP/4717753-09d6f9b0
    -- Stopped music on hold on SIP/4717753-09d6f9b0
AGI Tx >> 200 result=-1
  == Spawn extension (mainmenu, 4716596, 1) exited non-zero on 'SIP/4717753-09d6f9b0'
    -- Executing [h@mainmenu:1] Hangup("SIP/4717753-09d6f9b0", "") in new stack
  == Spawn extension (mainmenu, h, 1) exited non-zero on 'SIP/4717753-09d6f9b0'
agi no debug
AGI Debugging Disabled
The 'agi no debug' command is deprecated and will be removed in a future release. Please use 'agi debug off' instead.

Offline SARK devs

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Re: Voicemail Disable The person at extension
« Reply #9 on: January 05, 2009, 06:29:21 PM »
OK....

At the linux console do...

db selintra-work setprop global MTIME OFF
db selintra setprop global MTIME OFF

Then go to globals panel and do a commit.

Then run your call again and let us know what happens (send another trace, - even if it is sucessful).

Best

S

Offline Graham

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Re: Voicemail Disable The person at extension
« Reply #10 on: January 05, 2009, 06:42:53 PM »
Thanks that fixed it, run into another issue so I've attached the log.

Did what you said with the IVR *+extnum, under LEAVE VOICEMAIL I select *+extnum.

I then call and press option one to leave a message and it kills the call, if I call *+extnum my self it puts me through.

Connected to Asterisk 1.4.21.1 currently running on server-1 (pid = 8312)
Verbosity is at least 4
server-1*CLI> agi debug yesAsterisk 1.4.21.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.21.1 currently running on server-1 (pid = 8312)
Verbosity is at least 4
server-1*CLI> agi debug yes
Usage: agi debug
       Enables dumping of AGI transactions for debugging purposes
    -- Executing [4716596@mainmenu:1] AGI("SIP/4717753-09d7b9e8", "selintra|Inbound|4716596") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Wait) Options: (1)
    -- AGI Script Executing Application: (Background) Options: (silence/1)
    -- <SIP/4717753-09d7b9e8> Playing 'silence/1' (language 'en')
    -- AGI Script Executing Application: (Wait) Options: (1)
    -- Playing 'usergreeting2002' (escape_digits=12) (sample_offset 0)
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/4717753-09d7b9e8' status is 'UNKNOWN'
    -- Executing [h@mainmenu:1] Hangup("SIP/4717753-09d7b9e8", "") in new stack
  == Spawn extension (mainmenu, h, 1) exited non-zero on 'SIP/4717753-09d7b9e8'
agi debug no
Usage: agi debug
       Enables dumping of AGI transactions for debugging purposes

Offline SARK devs

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Re: Voicemail Disable The person at extension
« Reply #11 on: January 05, 2009, 08:29:31 PM »
OK...

Can you send (or post here) the result of

db selintra show global

send to admin@aelintra.com if you don't want to post it.
What is the name of the cluster you are using?

also - can you run the second trace again but this time with agi debug?  ('agi debug on' won't work...  just type agi debug)

Thx

S

Offline Graham

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Re: Voicemail Disable The person at extension
« Reply #12 on: January 05, 2009, 08:50:21 PM »
Asterisk 1.4.21.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.21.1 currently running on server-1 (pid = 8312)
Verbosity is at least 4
server-1*CLI> agi debug
AGI Debugging Enabled
    -- Executing [4716596@mainmenu:1] AGI("SIP/4717753-09d7d2b0", "selintra|Inbound|4716596") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/4717753-09d7d2b0
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1231184291.74
AGI Tx >> agi_callerid: Hide_Number
AGI Tx >> agi_calleridname: Hide_Number
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 4716596
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: mainmenu
AGI Tx >> agi_extension: 4716596
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE RINGDELAY
AGI Tx >> 200 result=1 (0)
AGI Rx << GET VARIABLE FAXDETECT
AGI Tx >> 200 result=1 (2)
AGI Rx << GET VARIABLE CALLRECORD2
AGI Tx >> 200 result=1 (None)
AGI Rx << GET VARIABLE LTERM
AGI Tx >> 200 result=1 (NO)
AGI Rx << SET VARIABLE MOH ""
AGI Tx >> 200 result=1
AGI Rx << SET CALLERID Hide_Number
AGI Tx >> 200 result=1
AGI Rx << DATABASE GET "STAT" "OCSTAT"
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE REMOTENUM "4716596"
AGI Tx >> 200 result=1
AGI Rx << GET VARIABLE MTIME
AGI Tx >> 200 result=1 (OFF)
AGI Rx << GET VARIABLE IFTIME(13:00-09:00|mon|*|*?CLOSED:OPEN)
AGI Tx >> 200 result=1 (CLOSED)
AGI Rx << GET VARIABLE VOICEINSTR
AGI Tx >> 200 result=1 (NO)
AGI Rx << DATABASE GET "STAT" "IVRSTAT"
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE OPEN "NO"
AGI Tx >> 200 result=1
AGI Rx << EXEC Wait 1
    -- AGI Script Executing Application: (Wait) Options: (1)
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE SYSOP
AGI Tx >> 200 result=1 (1001)
AGI Rx << EXEC Background silence/1
    -- AGI Script Executing Application: (Background) Options: (silence/1)
    -- <SIP/4717753-09d7d2b0> Playing 'silence/1' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE CLUSTEROP "1001"
AGI Tx >> 200 result=1
AGI Rx << EXEC Wait 1
    -- AGI Script Executing Application: (Wait) Options: (1)
AGI Tx >> 200 result=0
AGI Rx << STREAM FILE usergreeting2002 "12" 0
    -- Playing 'usergreeting2002' (escape_digits=12) (sample_offset 0)
AGI Tx >> 200 result=49 endpos=45440
AGI Rx << GET VARIABLE SYSOP
AGI Tx >> 200 result=1 (1001)
AGI Rx << SET CALLERID Hide_Number
AGI Tx >> 200 result=1
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/4717753-09d7d2b0' status is 'UNKNOWN'
    -- Executing [h@mainmenu:1] Hangup("SIP/4717753-09d7d2b0", "") in new stack
  == Spawn extension (mainmenu, h, 1) exited non-zero on 'SIP/4717753-09d7d2b0'
agi no debug
AGI Debugging Disabled