how to remove the custom voicemail message in asterisk?

Can you explain what you mean. What is the 'default message' you refer to?

Normally it will play 'the nerson at extension xxx is unavailable...". If you record your name then that is played instead of the extension number.
You can record your own complete unavailable and busy messages.
You can also record a temporary message which is played if it is set and is designed for when you are out of the office etc...
There is also a message which can be played at the end which sais something like 'press # to re-record your message' which can be turned on and off in the dialplan.

the message is

"Please leave your message after the tone, when done hang up or press the pound key."

You can record your own complete unavailable and busy messages.
Ive done this and the message above plays straight after.

ive jsut dome some test with messages via the phone.

This seems to be played after the temporary message.

This is the code that seems to get activated.

exten=s,4,Goto(default|6001|1)

It looks like someone has replaced the message that gets played after the temporary one with a completely different one for some reason. You will need to look through your dialplan to see where Voicemail() is called and insert the 's' character.
i.e Instead of Voicemail(u6001) you would have Voicemail(su6001).

here is my current plan. there are no reference to voicemail(u6001) anywhere.

comment=Welcome
alias_exten=7000
include=default
include=default
exten=0,1,Goto(voicemenu-custom-1|s|1)
exten=1,1,Goto(voicemenu-custom-2|s|1)
exten=2,1,Goto(voicemenu-custom-3|s|1)
exten=s,1,Wait(1)
exten=s,2,Answer
exten=s,3,Background(record/Welcome_message)
exten=s,4,WaitExten(12)

comment=FM
alias_exten=6001
exten=s,1,Goto(ringroups-custom-2,s,1)
exten=s,2,Wait(2)
exten=s,4,Goto(default|6001|1)
exten=t,1,Goto(voicemenu-custom-1|s|1)
exten=i,1,Playback(invalid)

They are jumping to voicemail-custom so you will need to look through all of them to see where the Voicemail() function is called.

here are all of the references to voicemail.

exten=s,n,Voicemail(6001,b)
exten=6050,1,VoiceMailMain(s6001)

 [macro-stdexten] - I dont think these are being used.
  exten=s-NOANSWER,1,Voicemail(${ARG1},u)
  exten=s-BUSY,1,Voicemail(${ARG1},b)
  exten=s-BUSY,2,Goto(default,s,1)

hmm two different ways of callign voicemail() and both of them technically depreciated in asterisk 1.4.
I assume you are running asterisk 1.4?

> exten=s,n,Voicemail(6001,b)
This will go to voicemail of extension 6001, play the busy messages and then the standard recording afterwards. This is what you say you are experiencing. Try changing it to :-
exten=s,n,Voicemail(6001|sb)

> exten=6050,1,VoiceMailMain(s6001)
This causes voicemail to be left in voicemail mailbox 6001. There should be no prompt whatsoever as a busy or unavailable is not specified and s is used to disable the standard message. Again this format is depreciated in asterisk 1.4 so you should be using :-
exten=6050,1,VoiceMailMain(6001|s)

When something is depreciated it means that it will work but you may get a warning in the logs. Hoever in the next major version (1.6) that format will be removed so it is best to fix the problems now otherwise if you upgrade asterisk without fixing it work it will stop working.

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

ok Ive worked it out.

it was going to a ring group which had specified to go to an extension of 6001.

which had my personal message plus the defaults.

so now it works. but the issue is now that the phone does not ring.

What im after is the phones to ring for 20 seconds (waiting for me to answer) then go to Voicemail(su6001)

Can you post the dialplan for the ring group and also the output to 'asterisk -r -vvv' which is displayed when the group is called.

asterisk -r -vvv
Asterisk Business Edition 1.1.1 (sx00i 1.1.1), Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Thank you for using Business Edition. This Software is provided by Digium Inc
under license. Please refer to the license agreement provided with the Software.
===============================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.1.1 (sx00i 1.1.1) currently running on asterisk (pid = 329)
No entry for terminal type "xterm";
using dumb terminal settings.
Verbosity is at least 3
 
    -- Starting simple switch on 'Zap/5-1'
 
    -- Executing [s@DID_trunk_1:1] ExecIf("Zap/5-1", "0|SetCallerPres|unavailable") in new stack
    -- Executing [s@DID_trunk_1:2] ExecIf("Zap/5-1", "0|Set|CALLERID(all)=unknown <0000000>") in new stack
    -- Executing [s@DID_trunk_1:3] Goto("Zap/5-1", "ringroups-custom-1|s|1") in new stack
    -- Goto (ringroups-custom-1,s,1)
    -- Executing [s@ringroups-custom-1:1] NoOp("Zap/5-1", "RINGGROUP") in new stack
    -- Executing [s@ringroups-custom-1:2] Dial("Zap/5-1", "Zap/1&SIP/6001|2|i") in new stack
 
    -- Called 1
    -- Called 6001
 
    -- Zap/1-1 is ringing
 
    -- SIP/6001-00362a80 is ringing
 
    -- Zap/1-1 is ringing
 
[Aug 11 21:50:59] WARNING[3103]: chan_zap.c:4183 zt_handle_event: Didn't finish Caller-ID spill.  Cancelling.
    -- Zap/1-1 is ringing
 
    -- Nobody picked up in 2000 ms
    -- Hungup 'Zap/1-1'
    -- Executing [s@ringroups-custom-1:3] Goto("Zap/5-1", "voicemenu-custom-1|s|1") in new stack
    -- Goto (voicemenu-custom-1,s,1)
    -- Executing [s@voicemenu-custom-1:1] Wait("Zap/5-1", "1") in new stack
 
    -- Executing [s@voicemenu-custom-1:2] Answer("Zap/5-1", "") in new stack
    -- Executing [s@voicemenu-custom-1:3] BackGround("Zap/5-1", "record/Welcome_message") in new stack
    -- <Zap/5-1> Playing 'record/Welcome_message' (language 'en')
 
  == CDR updated on Zap/5-1
    -- Executing [1@voicemenu-custom-1:1] Goto("Zap/5-1", "voicemenu-custom-2|s|1") in new stack
    -- Goto (voicemenu-custom-2,s,1)
    -- Executing [s@voicemenu-custom-2:1] VoiceMail("Zap/5-1", "su6001") in new stack
    -- <Zap/5-1> Playing '/var/spool/asterisk/voicemail/default/6001/temp' (language 'en')
 
    -- <Zap/5-1> Playing 'beep' (language 'en')
 
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/6001/tmp/voYnrn format: wav, 0x360fa8
 
    -- User hung up
 
  == Spawn extension (voicemenu-custom-2, s, 1) exited non-zero on 'Zap/5-1'
    -- Hungup 'Zap/5-1'
 
asterisk*CLI>

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comment=Welcome
alias_exten=7000
include=default
exten=s,1,Wait(1)
exten=s,2,Answer
exten=s,3,Background(record/Welcome_message)
exten=s,4,WaitExten(12)
exten=0,1,Goto(voicemenu-custom-1|s|1)
exten=1,1,Goto(voicemenu-custom-2|s|1)
exten=2,1,Goto(voicemenu-custom-3|s|1)


comment=FM
alias_exten=6001
exten=0,1,Goto(voicemenu-custom-1|s|1)
exten=s,1,Voicemail(su6001)


comment=SWH
alias_exten=6003
exten=s,1,Voicemail(su6001)
exten=0,1,Goto(voicemenu-custom-1|s|1)

comment=ringroups-custom-1
gui_ring_groupname=ALLPHONES
exten=s,1,NoOp(RINGGROUP)
exten=s,n,Dial(Zap/1&SIP/6001,2,i)
exten=s,n,Goto(voicemenu-custom-1|s|1)

comment=Default
exten=6050,1,VoiceMailMain(su6001)
exten=7000,1,Goto(voicemenu-custom-1|s|1)
exten=6001,1,Goto(voicemenu-custom-2|s|1)
exten=6003,1,Goto(voicemenu-custom-3|s|1)

so all im after is...

comment=FM
alias_exten=6001
exten=0,1,Goto(voicemenu-custom-1|s|1)
RING ALL PHONES FOR 15 SECONDS - waiting for me to pickup.
exten=s,1,Voicemail(su6001)

> exten=s,n,Dial(Zap/1&SIP/6001,2,i)
You are only ringing the phones for 2 seconds and it is likely that they wont even have time to respond and start ringing in that time. As you want 15 seconds you should be using :-
exten=s,n,Dial(Zap/1&SIP/6001,15,i)

ive made the change but its stil not ringing the phones.

All it does now, is plays the main menu (welcome)

then when I press 1 it goes straight to the mailbox message. I need a 15 second delay/ring all phones here.

Can you post the 'asterisk -r -vvv' output again because the earlier one you posted did show it ringing the phones.

ive called the main number.

listened to the main message in welcome, then pressed 1. and immediately it went to the temporary greeting that I created through the phone.

I would like to have a 15 second period where the phones ring so I can answer it.

root:~> asterisk -r -vvv
Asterisk Business Edition 1.1.1 (sx00i 1.1.1), Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Thank you for using Business Edition. This Software is provided by Digium Inc
under license. Please refer to the license agreement provided with the Software.
===============================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.1.1 (sx00i 1.1.1) currently running on asterisk (pid = 335)
No entry for terminal type "xterm";
using dumb terminal settings.
Verbosity is at least 3
 
    -- Starting simple switch on 'Zap/5-1'
 
    -- Executing [s@DID_trunk_1:1] ExecIf("Zap/5-1", "0|SetCallerPres|unavailable") in new stack
    -- Executing [s@DID_trunk_1:2] ExecIf("Zap/5-1", "0|Set|CALLERID(all)=unknown <0000000>") in new stack
    -- Executing [s@DID_trunk_1:3] Goto("Zap/5-1", "ringroups-custom-1|s|1") in new stack
    -- Goto (ringroups-custom-1,s,1)
    -- Executing [s@ringroups-custom-1:1] NoOp("Zap/5-1", "RINGGROUP") in new stack
    -- Executing [s@ringroups-custom-1:2] Dial("Zap/5-1", "Zap/1&SIP/6001&SIP/6002|1|i") in new stack
 
    -- Called 1
 
    -- Called 6001
 
[Aug 11 22:55:43] WARNING[743]: chan_sip.c:2964 create_addr: No such host: 6002
[Aug 11 22:55:43] WARNING[743]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
    -- Zap/1-1 is ringing
 
    -- SIP/6001-01584004 is ringing
 
    -- Zap/1-1 is ringing
 
[Aug 11 22:55:44] WARNING[743]: chan_zap.c:4183 zt_handle_event: Didn't finish Caller-ID spill.  Cancelling.
    -- Zap/1-1 is ringing
 
    -- Nobody picked up in 1000 ms
    -- Hungup 'Zap/1-1'
    -- Executing [s@ringroups-custom-1:3] Goto("Zap/5-1", "voicemenu-custom-1|s|1") in new stack
    -- Goto (voicemenu-custom-1,s,1)
    -- Executing [s@voicemenu-custom-1:1] Wait("Zap/5-1", "1") in new stack
 
    -- Executing [s@voicemenu-custom-1:2] Answer("Zap/5-1", "") in new stack
    -- Executing [s@voicemenu-custom-1:3] BackGround("Zap/5-1", "record/welcome2") in new stack
 
    -- <Zap/5-1> Playing 'record/welcome2' (language 'en')
 
  == CDR updated on Zap/5-1
    -- Executing [1@voicemenu-custom-1:1] Goto("Zap/5-1", "voicemenu-custom-5|s|1") in new stack
    -- Goto (voicemenu-custom-5,s,1)
    -- Executing [s@voicemenu-custom-5:1] VoiceMail("Zap/5-1", "su6001") in new stack
[Aug 11 22:55:52] WARNING[743]: app_voicemail.c:7185 vm_exec: Prefixing the mailbox with an option is deprecated ('su6001').
[Aug 11 22:55:52] WARNING[743]: app_voicemail.c:7186 vm_exec: Please move all leading options to the second argument.
 
    -- <Zap/5-1> Playing '/var/spool/asterisk/voicemail/default/6001/temp' (language 'en')
 
    -- <Zap/5-1> Playing 'beep' (language 'en')
 
    -- Recording the message
 
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/6001/tmp/q4QrRo format: wav, 0x153c950
 
    -- User hung up
 
  == Spawn extens

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-- Executing [s@ringroups-custom-1:2] Dial("Zap/5-1", "Zap/1&SIP/6001&SIP/6002|1|i") in new stack
You are only ringing the phones for 1 second not 15.

[Aug 11 22:55:43] WARNING[743]: chan_sip.c:2964 create_addr: No such host: 6002
[Aug 11 22:55:43] WARNING[743]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Phone 6002 is not registered with the PBX so that phone cannot ring.

the 1 second is the first ring to get to the main menu.
Setting it to 15 seconds would just be too long.

On the first ring it goes to welcome in 1 second > plays the welcome message.
then when I press 1 for sales it goes to FM menu and straight into voicemail. it is here i require the 15 seconds

The log you posted calls voicemail-custom-1. It looks similar to 'Welcome' that you posted earlier but is oviously different.
Can you post this config as it is the one you are actually using.

Im going to have to go to bed.

it is so late over here. thankyou so much for your help. i know it must be frustrating.

include=default
comment=Welcome
alias_exten=7000
exten=s,1,Wait(1)
exten=s,2,Answer
exten=s,3,Background(record/welcome2)
exten=s,6,WaitExten(10)
exten=0,1,Goto(voicemenu-custom-1|s|1)
exten=1,1,Goto(voicemenu-custom-5|s|1)
exten=2,1,Goto(voicemenu-custom-3|s|1)
exten=t,1,Goto(voicemenu-custom-1|s|1)
exten=i,1,Playback(invalid)
 
comment=FM
alias_exten=6001
exten=0,1,Goto(voicemenu-custom-1|s|1)
exten=s,1,Voicemail(su6001)
 
comment=SWH
alias_exten=6003
exten=s,1,Voicemail(su6001)
exten=0,1,Goto(voicemenu-custom-1|s|1)
 
comment=ringroups-custom-1
gui_ring_groupname=All Phones
exten=s,1,NoOp(RINGGROUP)
exten=s,n,Dial(Zap/1&SIP/6001,15,i)
exten=s,n,Goto(voicemenu-custom-1|s|1)
 
comment=ringroups-custom-2
gui_ring_groupname=Business Hours
exten=s,1,NoOp(RINGGROUP)
exten=s,n,Dial(Zap/1&SIP/6001,5,i)
exten=s,n,Voicemail(6001,b)
 
 
(not sure if this is needed)
exten=6050,1,VoiceMailMain(s6001)
exten=7000,1,Goto(voicemenu-custom-1|s|1)
exten=6003,1,Goto(voicemenu-custom-3|s|1)
exten=6001,1,Goto(voicemenu-custom-5|s|1)

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Digium logged in a fixed a bug with my version of asterisk business edition.

All working now. some of your points helped me in other areas.

thanks.