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5. Asterisk Voicemail

VoiceMail is used to leave a message if no one is answering your call. The configuration in Asterisk is again in /etc/asterisk and the file is voicemail.conf.

You can declare the mailbox in the default mailbox context [default] or create others. Note that the mailbox contexts and those in extensions.conf have no relation in between. They are just some kind of separators in two different files.

The command looks like this:

mailbox_number => password, name, email

mailbox_number is the number you use in extension.conf for VoiceMail() command and to register a user in sip.conf or iax.conf
password is the pass used to register a user in sip.conf or iax.conf
name is the name which to be associated with the mailbox
email is where a notification for the voicemail will come

Ex. (voicemail.conf)

[mb_tutorial]
777 => 1212, ivan, ivan@asteriskguru.com

here I create mailbox context named mb_tutorial, mailbox number 777 with password 1212, owned by ivan with email ivan@asteriskguru.com


When dialing any user you can leave message at this mailbox

VoiceMail(mailbox_number@context)
Ex. (extensions.conf)

exten => 1234,1,Dial(SIP/ivan, 30)
exten => 1234,2,VoiceMail(777@mb_tutorial)
exten => 1234,3,PlayBack(vm-goodbye)
exten => 1234,4,HangUp()



This example will try dialing SIP user ivan at number 1234 for 30 seconds and after this if nobody picks up the extension with next priority level is to be executed i.e. the asterisk will open the 777 mailbox at context mb_tutorial. Then playback will run and the asterisk will hang you up. PlayBack (vm-goodbye) plays the vm-goodbye file which has to be in /var/lib/asterisk/sounds/.

The recorder voicemail messages are recorder in /var/spool/asterisk/voicemail/<context>/<mailbox>/INBOX/ so here the path will be /var/spool/asterisk/voicemail/mb_tutorial/777/INBOX/.

To listen to your mailbox you have to use the VoiceMailMain command.

Synopsis : VoiceMailMain(mailbox@context)

Here is what you can do with your mailbox using VoiceMailMain.

1 Old Messages

3 Advanced options

1 Send reply
2 Call back
3 Envelope
4 Outgoing call
5 Leave message
* Return to main menu

4 Play previous message
5 Repeat current message
6 Play next message
7 Delete current message
8 Forward message to another mailbox
9 Save message in a folder
* Help; during msg playback: Rewind
# Exit; during msg playback: Fastforward

2 Change folders
3 Advanced options
0 Mailbox options

1 Record your unavailable message
2 Record your busy message
3 Record your name
4 Change your password
* Return to the main menu

* Help
# Exit

While you listen to any recorded voicemail message you can user the following keys for navigation:

# to fastforward
* to rewind
Note :'#' and '*' keys only work when the message is in the process of being played back.

Ex. exten => 9999,1,VoiceMailMain(777@mb_tutorial)


dialing 9999 you will enter the 777 mailbox after
typing the correct password for the mailbox (1212).


Here is how our VoiceMail example looks like:


1. Create voicemail context and mailbox at voicemail.conf

voicemail.png

We created mailbox context [mb_tutorial] and also mailbox number 777 with password 1212 for user ivan with email ivan@asteriskguru.com.

2. SIP users in sip.conf


If you are using voip hardphone which supports voicemail messaging, then you can check your mailbox from the phone. As the harphones communicate with asterisk through session initiation protocol (SIP) this feature have to be adjusted in sip.conf. When you receive a message MWI indication (message waiting indication) will appear on the phone.

3. We will also format a little extensions.conf



exten => 1234,1,Dial(SIP/ivan, 45)
Dial user ivan with SIP for 45 sec

exten => 1234,2,VoiceMail(777@mb_tutorial)
If nobody answers or the line is busy or congested you enter VoiceMail to leave message on 777

exten => 1234,3,PlayBack(vm-goodbye)
When done asterisk plays goodbye message

exten => 1234,4,Wait(2)
Wait 2 more seconds

exten => 1234,5,HangUp()
Asterisk closes the connection



 
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Nick (kmvd68 at motorola dot com)
28 April 2009 15:22:27
I would like to know if the Asterisk platform can be integrated to work with a Nortel MSC (Mobile Switching Center)? If so,
what would be the software/hardware requirements?

Thanks
shalu (shalu dot dhamija at rancoretech dot com)
03 September 2008 15:10:55
Hi,
I am using asterisk from the last few days but i did not find the code for SUBSCRIPTION to Voicemail service (i.e.SIP SUBSCRIBE Messages for Voicemail). It would be a great help if somebody can tell me the filename containing the code for the same.

Thanks and Regards,
Shalu Dhamija
marco (m dot criscuolo at elmansrl dot it)
03 September 2008 09:08:22
I'm have a great problem. I don't succeed in sending e-mail through voicemail. I can register messages on the mailbox specified in voicemail.conf but Asterisk doesn't send notification e-mails. I have ssmtp as MTA, anybody can help me?
Josh K (it at hhtcfs dot com)
21 July 2008 15:38:31
Just like so many else, I'm having a problem that some of my phones aren't getting voicemail. I check in the directory and new one's (that I'm leaving myself) aren't even being saved. I have 2 people w/ this problem out of 40. They are set up in the config files just the same as everybody else! Please help
czervik (dave at castlewire dot com)
16 July 2008 18:40:50
Voice mail prompts callers to hang up or dial "#" to end their message. If you dial "#" the call hangs up and then dials something else and the standard "The call you made can not be completed as dialed...". Any idea why this happens?
Sheldon (eaglefish55-apricots at yahoo dot ca)
08 July 2008 05:33:38
I've got a weird problem.
I'm trying to identify the source of a message I hear when I'm calling the incoming DID line I've just started up.

First there's no ring on the number
Then there's this female telephone company type voice that goes
"One moment please, the person at extension 1001 is not available..."

I'm wondering if that is a default voicemail box set up by Asterisk.

The guys I've got the line from may be using Asterisk. What's weird is that this is a very intermittent problem with my application, a sophisticated telephone answering system working about half the time. The problem only seems to affect my line.

I got the idea it might be asterisk after seeing a reference to asterisk at my did supplier web site and a google search.

Thanks for any help
Rigoberto Lorenzana (rlorenzana at logitel dot com dot mx)
06 June 2008 23:40:48
I'm looking for a voice mail server, which can be connected even Sotfswicht Galery, 1000 user and 23 user ports.
rlorenzana@logitel.com.mx
Binh tran (binh_tranph at yahoo dot com)
05 May 2008 18:23:32
thanks very much...
Mike Munroe (mmunroe at rjburnside dot com)
02 April 2008 21:56:23
I have 10 Asterisk servers running I would like to be able to set up dundi lookup for forwarding or sending Vmail

every user has a diff ext number and we use dundi to lookup and direct dial them but can only forward with mail with that server any ideas
tlc (Timothy dot L dot Cline at embarq dot com)
12 March 2008 19:43:00
when one extension on an FSO port calls another extemsion on a different FSO port and is forwarded to voice mail it is asking for the login to voice mail box.
I have configured 6050, by default, for mailbox access.

Should there be another number configured for LEAVING voice mail?
mahya (mahyash2001 at yahoo dot com)
09 February 2008 12:45:38
Hi,
When I use Voicemail function, there is a default system greeting
before voicemail recording. Is it possible to change that greeting?
How? Actually I want to change the sequence of sound file that played for greeting.Pleaze send your solution to my mail box.
Thanks
sarkash (sarkash0070 at yahoo dot com)
12 December 2007 12:59:32
hi ..

Where is the answers??
Polar Star (mcoghlan at myrealbox dot com)
27 November 2007 06:46:00
Thanks Jez! You the man.
Mardonio Sanchez (mar2s at hotmail dot com)
19 August 2007 05:54:25
Recently I installed the new version of FreePBX 2.2.3. Everything is working O.K. but when I want to hear my VM I have the following error:
-- Incorrect password '1111110022' for user '11102' (context = default).

Any advice is appreciated.
naning (naning dot purisima at gmail dot com)
17 August 2007 05:21:11
Hi All!

I have a concern on my Asterisk PBX, since it is wokrs fine, whe i logged on to the Asterisk console, i have noticed a warning message, WARNING[1920]: chan_sip.c:6545 determine_firstline_parts: Bad request protocol. I cannott figure out that message where is it came from. Anyone can help me?
Arfeen (arfeenster at gmail dot com)
25 May 2007 08:03:21
my VM is working fine... how could I change the filename of message ? currenly it sets like 'msg000x' . how do I customize that?
Rick (rick dot f at markerstudio dot com)
22 May 2007 02:45:59
Re Voicemail to Email issues. The problem that Asterisk cant send emails is due to the setup of your linux server. I use Debian so had to run the config for mail. Then set it to allow sending mail to an smtp host (off by default) If you dont so this then you will not get any errors from Ast, but you also will not recieve mails.
Asha Rose (asha-pg6 at iiitmk dot ac dot in)
16 May 2007 14:01:48
the thing is that..i tried the voicemail option and it is working perfectly..now the trouble is with sending it as a notification to the user's mail id's. The users are not receivin the mail notifications in their mail account.so how to make it work?
How to do the send mail options?
Atauar (atauar at yahoo dot com)
21 April 2007 22:54:35
Hi,
my MV is working fine but it can't send mail.
can anybody help me ? what is tha problem?
Rose (asha-pg6 at iiitmk dot ac dot in)
02 April 2007 06:31:21
hi all
I have problem when i try to log in to the voicemail box.eventhough i gave the right password it is saying that the password is incorrect. why?

in extensions.conf

[voicemail]
exten => 8500,1,VoiceMailMain

[friends]
exten => 333,1,Dial(IAX2/asha,10)
exten => 333,2,VoiceMail(333@voicemail)

exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
include =>voicemail

In voicemail.conf

[voicemail]
333 => 3333,asha,asha-pg6@iiitmk.ac.in
avinash (avinash_alb at yahoo dot com)
30 March 2007 09:31:57
can u tell me how to configure mediant 1000 audio codes? because im not able to configure and u just tell me step by step so i can follow ur advice
Ajit Kumar (kumarsujitsingh at yahoo dot com)
29 December 2006 08:35:21
Hi Support Team,

Myself ajit kumar working on asterisk for last 1 year.I just wanted to know if I implement iptables on asterisk server,What will be the demarites and any idea how to implement it.
With Warm Regards
Ajit Singh
+91 9910705847
Jim (xapmat at mac dot com)
30 November 2006 04:58:13
I keep getting the 'incorrect password' message when I try to enter in the password for a mailbox. When looking at the console I get this:
Incorrect password '11' for user '101' (context = VoicemailContext)
or variations on it '1' or '111'
The password is 1111. It's like it's not getting all of it. Any ideas?
Boho (gxb2002 at gmail dot com)
24 October 2006 07:47:11
1. exten => 1234,1,Dial(SIP/ivan, 30)
2. exten => 1234,2,VoiceMail(777@mb_tutorial)
3. exten => 1234,3,PlayBack(vm-goodbye)
4. exten => 1234,4,HangUp()
I have a problem that between the line2 and line3, how long will the line2's program run?And how the line3 know the line2's program has end?
Thanks for reading my problem with the poor English.:-(
Emile (emilrawk at hotmail dot com)
22 October 2006 21:32:27
check previous post

voicemail.conf
[bv_home]
8383 => 1234, emile, emile@slackterisk.com

extensions.conf
[home]
exten => 8383,1,Dial(SIP/emile, 15)
exten => 8383,2,VoiceMail(8383@bv_home)
exten => 8383,3,PlayBack(vm-goodbye)
exten => 8383,4,HangUp()


sip.conf
[emile]
type=friend
username=emile
secret=1234
host=dynamic
context=home
mailbox=8383@bv_home
Emile (emilrawk at hotmail dot com)
22 October 2006 21:30:52
hi here is what i got.
When i dial ext 8383, after 15 seconds i leave a message.
when i dial ext 500 for voicemailmain to ear the message left, i always get login incorrect message. it looks like only the extension in default is working. The mail doesnt look like it see context [bv_home].

CLI pull an error cant register user .
Jonathan (jonsol at gmail dot com)
19 October 2006 20:28:57
I hope someone could help me.
I have 2 telephones connected at the same line, and that line is connected to my asterisk server (a computer with two ports, one FXO port and one FXS port).

I can call and put some wait time and then, if no body answer then the voicemail gets in action.

But I have 2 problems on that:
#1 - If I answer a call at one of the others telephones, asterisk seems not to be noticed and after the wait time answer and record the message even when I already had picked up the incoming call, don't know if there exist some command to prevent that or if I missed some configuration issue.

#2 - If I'm using the line and then some one else call, I need to run the voicemail, but asterisk seems not heared that other incoming call, I don't see the usual: -- Starting simple switch on 'Zap/4-1' at the CLI.

Thanks
Jonathan (jonsol at gmail dot com)
19 October 2006 20:19:50
I think you can change the message running the VoiceMailMain program:
exten => 1000,1,VoiceMailMain(exten@context)

watch this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoicemailMain

the are the options you have, one of them is change the messages
ross (ross_beer at hotmail dot com)
17 October 2006 19:52:26
Does anyone know how to change the default busy message, so that it says the telephone number called instead of the mailbox number?

Do i need to manualy make a prompt and then use voicemail application to handle recoring etc?

ie.
playback(message)
voicemail(boxnumber)
hangup
John_Z (gtr_1000 at yahoo dot com)
10 October 2006 07:55:25
Have been playing around last night with the settings in the AudioCodes.
AudioCodes hangs up due to the broken_connection setting this is defaulted to 10 seconds. You can disable this setting, my problem with that is that I don't get a disconnect upon hangup and end up with 3 minutes voicemail )max message setting)
Have been playing around with the reverse_polarity and current_disconnect but haven't found a solution as of yet.

Also set the setting of the RTP stream to 5 msec in stead of the default setting, but no luck
Looks like the AudioCodes hangs up because it doesn't receive anything from Asterisk.

John
John_Z (nospam at nospam dot com)
08 October 2006 14:02:58
Did anyone overcome the 10 seconds limit with the AudioCodes??

In the trace I see the following, calling from ther outside world:
-- SIP/voipbuster-fdf2 is making progress passing it to SIP/2102-056d
-- Executing Dial("SIP/mp108in-a58d", "SIP/2101|20|r") in new stack
-- Called 2101
-- SIP/2101-eb05 is ringing
-- Nobody picked up in 20000 ms
-- Executing VoiceMail("SIP/mp108in-a58d", "u2101") in new stack
-- Playing '/var/spool/asterisk/voicemail/default/2101/unavail' (language 'en')
-- SIP/voipbuster-fdf2 answered SIP/2102-056d
-- Attempting native bridge of SIP/2102-056d and SIP/voipbuster-fdf2
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/2101/INBOX/msg0005 format: wav, 0x94ef9a0
Oct 8 13:51:24 WARNING[10230]: app.c:644 ast_play_and_record: No audio available on SIP/mp108in-a58d??
-- User hung up
== Spawn extension (internal, 2101, 2) exited non-zero on 'SIP/mp108in-a58d'

Somehow it looks like the AudioCodes hangs up.
Anyone an idea??
Kotek (listy at ukotka dot com)
03 October 2006 08:27:23
Regarding the incorrect password, the example working for me on AstLinux is:
exten => 2468,1,VoicemailMain(@mb_tutorial)

This enters the correct context and allows you to specify the mailbox number and passwrd. All examples I have found were showing context without @ symbol which was the cause of incorrect login. HTH
keyur patel (keyur4wish at gmail dot com)
11 September 2006 13:22:09
hi

Can any body give me step by step guide for asterisk.which service is responsible for loading asterisk. please ...................

regards,
keyur
Aristide (astnic20 at yahoo dot fr)
25 August 2006 13:09:31
if i put as e-mail address : root@localhost where do asterisk send the voice messages?
i want to sent notification to the user when receiving a voice message.what do i do?

thanks for ur help
Ross (tordah at gmail dot com)
25 August 2006 11:16:44
Forwarding messages to email is simple, just set it in voicemail.conf. I don't think you need to set anything else on your server for it to send emails out - the beauty of linux.

Example
200 => 200 , Bob, bob@hotmail.com, bob@asteriskserver

Each voicemail message should be sent to bob@hotmail.com.
sara (saggit_16 at yahoo dot com)
23 August 2006 09:55:54
how can i use atmail to record voice mail and then play it using a normal sip phone all this being done through asterisk
please reply a.s.a.p
SkyRanger (skyr at inbox dot ru)
10 August 2006 10:48:04
Hi to all.
I have problem with logining in voice mail...

Here sip.conf

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
context = default ; Send unknown SIP callers to this context
srvlookup=yes

[2001]
type=friend
host=dynamic
username=2001
secret=1234
canreinvite=no
nat=yes
context=phones
dtfmode=rfc2833
allow=all
mailbox=2001@phones

and here extensions.conf:

[general]
static=yes
writeprotect=no

[globals]
JF=SIP/2001
JF1=SIP/2002

[defaults]

[phones]

exten => 2001,1,Dial(${JF},15)
exten => 2001,2,Voicemail(u${JF}@${CONTEXT})
exten => 2001,102,Voicemail(b{JF}@${CONTEXT})

exten => 2002,1,Dial(${JF1},15)
exten => 2002,2,Voicemail(u${JF1}@${CONTEXT})
exten => 2002,102,Voicemail(b{JF1}@${CONTEXT})

exten => 501,1,VoicemailMain

exten => 9999,1,VoicemailMain(2001)

exten => h,1,Hangup

exten => 500,1,Answer()
exten => 500,2,Playback(demo-echotest)
exten => 500,3,Echo
exten => 500,4,Playback(demo-echotest)
exten => 500,1,Hangup()

and now my part voicemail.conf:

[phones]
2001 => 9999,Sky Ranger,skyr@inbox.ru
2002 => 9999,Sky Ranger,skyr@inbox.ru

when i dial 501 or 9999 i hear prompt for mail box i type 2001 for password i type 9999 and then it says that login incorrect :(

what i do wrong????
jason (jasloan at adelphia dot net)
07 August 2006 10:11:50
There used to be an outcall application supported or in development for asterisk. I found remnants of it in a google search, since I am just sorta experimenting with asterisk for the fun of it I only have a sipura adapter and my standard analog phone. What I'm trying to do is setup an outcall feature that calls back the house phone's extension say every hour or so when there's new voicemail in the mailbox. However I can find no current documentation on this feature or proof that it even exists save the 2004 bounty at SineApps. Does anyone have any information on this feature or know where I can find some more info?

Thanks.
Pravin (dabhade_pravin at yahoo dot com)
25 July 2006 16:07:58
Hi
I am sucessfully able to configure asterisk with ambient md3200 card
as well as able use the basic epbx functionality . also I able to set the voicemail functionality .
when I use the voiceMailmain it request for the mailbox number and play the voice menu also it will detect correctly the number of message in mailbox . but when I press '1' to read the new message as per the voice menu it wont play the voice message . aslo I am not able to select the other options

but the files are created for that mailbox in /var/spool/asterisk/vm
John (junkoneus at yahoo dot com)
25 July 2006 12:36:59
HELP Please: Voicemail recording length issue. using asterisk and AudioCodes SIP and Cisco handsets...
If call from extension to extension then vmail recording length OK
If call from outside line to extension then vmail recording length only 10 sec.
austre (austre at hotmail dot com)
06 July 2006 23:21:24
hi !!! I am tryinto to send a voicemail message but i think that is recordes but when i enter to my mailvoice nothing is recorded, thank for you help !!!
Amit Goel (amitgoel74 at rediffmail dot com)
03 July 2006 16:28:17
Hello !! I just want to know how can we check voice mail messages in our inbox of hotmail or rediffmail box or like that. if we can not do this than how we can forwarded these voice messages to email address..
Regards

Amit Goel
uzair (uzairqazi18 at hotmail dot com)
09 June 2006 13:34:42
how can we check voice mail messages in our inbox of hotmail or like that.
sanjay marathe (SMarathe at trinityconvergence dot com)
07 June 2006 10:07:20
Hi, I would like to know if Asterisk supports SIP based voicemail . If yes, how do I register a SIP user with Asterisk for vociemail and how do I enable "message waiting indication " ?

thanks
rob (rossirj at slu dot edu)
16 May 2006 20:45:04
I am having a problem with a just installed AAH. When the VM picks up there is no sound (no attendent voice can be heard). Any ideas how to fix this?
joetke (zanhepoh at yahoo dot com)
14 May 2006 23:40:57
Thanks
Your tute helped me but sometimes your english confused me. English is not my mother tongue though. Is there anyone who could help you reformulate a few sentences to get them much clearer.
ever zalazar (zalazar at technologymaker dot com)
11 May 2006 18:26:40
Hi guys, there is some way to send a notification to a SIP EXPRESS ROUTER, when receive a voicemail?
samuel (levysamuel at hotmail dot com)
08 May 2006 14:48:13
How can I write a general extension to access the voice mail boxes
(not a VoiceMailMain for each user)
StasK (stas at wirewalk dot com)
01 May 2006 16:03:21
Jez
thanks for advice
worked like a charm
on both grandstream budgetone 100 and gxp 2000
Jez (jeremy at medialicious dot com)
19 April 2006 13:05:57
Just a quick tip for Grandstream users:

If, when accessing your voicemail, the automated voice speaks but the phone is not responding to any keys, do the following:

1) Load your web browser and enter the IP of the phone.
2) At the Grandstream Device Configuration screen, select Advanced Settings.
3) Change the "Send DTMF" from 'in-audio' to 'via SIP INFO'.
4) Update and reboot the phone and voicemail will work correctly.

Hope this helps someone out!
Raheel (raheel213 at yahoo dot com)
20 March 2006 09:27:09
can any one teel me how these voice messages will be forwarded to email address..
mbergen (mbergen at here dot ca)
04 March 2006 15:28:12
The @context tip was great.
Thanks.
andy (endocrantz at yahoo dot com)
26 January 2006 01:36:04
The above specifies the voicemail context "mb_tutorial" in both voicemail.conf and extensions.conf

As the voicemail is located in this special context asterisk VoicemailMain will only find it if you specify that context in extensions.conf:

exten => 2468,1,VoicemailMain(777@mb_tutorial)
exten => 2468,n,Goto(s,6)

Dialing 2468 will call the 777 voice mailbox now

Without a context specified asterisk only searches [default]
A few other tricks can be accomplished, See:
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain
ortega (ortega at codinet dot com dot mx)
13 November 2005 15:55:42
Try changing the value dtmfmode in sip.conf
borghart (borghart at gmx dot de)
06 November 2005 12:44:07
@Alex: i'm having the same problem: although i enter the correct password (asterisk -vvv shows stuff like "Sending dtmf: 42 (*), at ip.address") asterisk does not react to the input. do you know how to fix this?
Alex (blanc dot noire at gmail dot com)
17 October 2005 05:06:41
Yeap i read it... but still can't seem to figure out why i can't access it...

thanks for your help Lacho
Lacho (support at asteriskguru dot com)
13 October 2005 14:42:45
Hi Alex. Did you read our tutorial about the VoicemailMain application. Perhaps you can find there the solution. There you can find more detailed information about this application.
Alex (blanc dot noire at gmail dot com)
13 October 2005 06:55:00
Hi there... can dial the VoiceMailMain, but the server tells me to "please enter password". even though i enter the correct password for the particular mailbox, nothing happens...what could be the reason? anyway to rectify it?
athanase (webmaster at esgis dot tg)
08 September 2005 14:44:33
i think that is cool but if you can be more clear in the email that record the voice in the server that can be interesthing.
i dont no that if my linux acount is athanase@athanase, i can use it like my mail in my voicemail configuration .
please explane it to me.
balvinder (balvinderjassar at gmail dot com)
05 August 2005 14:13:03
helped me out newbees must try this
 
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    added 05/Nov/2007 10:24
  • Zoiper
    added 22/Oct/2007 17:53

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