Re: [asterisk-users] Voicemail Transcription with openai/whisper

[David Rebarchik]

I really love this idea. Thanks for sharing. I've been looking for a 
good way to provide this service to my customers. Hopefully this will 
work for me too.


Thanks,
Dave

On 11/27/2022 8:08 AM, Doug Lytle wrote:

Everybody,

I've recently discovered openai/whisper and have been trying in 
earnest to get this working with Asterisk for voicemail transcriptions 
(Currently using the NerdVittles script with IBM Watson)


https://github.com/openai/whisper

After spending several hours today, I've successfully integrated my 
home Asterisk 16 voicemail with Whisper.


Once I have followed the instructions for setting up an API server

https://blog.deepgram.com/how-to-build-an-openai-whisper-api/

Initially, I setup a quad core VM to test this with, but discovered 
that without a dedicated card for the inference that it was horribly 
slow.  So, I've set up testing on my desktop (Kubuntu 20) since I have 
an nVidia GTX 1060 installed.


For the integration with Asterisk, I'm using a slightly modified 
script from nerdvittles IBM Watson script


sendmailibm

That can be found on their website

https://nerdvittles.com/free-asterisk-voicemail-transcription-with-ibms-stt-engine/

I will probably find a low cost nVidia video card and get a stand 
alone Linux box running to handle this project.


If you're interested in the details, let me know.

Doug




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Re: [asterisk-users] Voicemail Transcription with openai/whisper

[Doug Lytle]

On 11/27/22 09:22, Greg Troxel wrote:

Thanks for posting.  As I'm running asterisk on a PC Engines apu2, I
don't need the details as it is obviously unworkable, but it's great to
see non-cloud progress.

Greg,

Just a note,

This would work if you have the API server running on a Linux x86 box.

Then Asterisk would be using curl and python to communicate with that 
API Linux box.


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Re: [asterisk-users] VoiceMail() stops dialplan processing

[Antony Stone]

On Monday 03 October 2022 at 14:14:54, Joshua C. Colp wrote:

> On Mon, Oct 3, 2022 at 9:11 AM Antony Stone <
> 
> antony.st...@asterisk.open.source.it> wrote:
> > Hi.
> > 
> > I have a dialplan which calls the VoiceMail() application, and I'm
> > getting the following behaviour:
> >  - if the inbound caller leaves a message, then presses #, and then
> > presses 1 to accept the recording, everything works as expected and the
> > dialplan continues processing after the line containing VoiceMail()
> > 
> >  - if the inbound caller leaves a message and then hangs up, the diaplan
> > simply stops executing with a message such as:
> > 
> > [2022-10-03 13:02:23.355976] pbx VERBOSE[19022][C-0556]: pbx.c:4413
> > in __ast_pbx_run: Spawn extension (RecordVM, 00xx74xx88xx90, 2) exited
> > non-zero on 'SIP/TrunkOne-0c12'
> > 
> > The subsequent commands in the dialplan do not get processed.
> 
> This is fundamentally how dialplan works. If a channel hangs up, then
> normal dialplan execution stops.

I suppose that fits other situtations, yes.

> > Can anyone suggest either why this would happen and how to get the
> > dialplan to continue processing under all circumstances, or at least how
> > to investigate futher what is causing this to happen?
> > 
> > I'm sure that leaving a message and hanging up the call should be valid
> > because that's what the default greeting message tells people they can
> > do.
> 
> It is. If you're needing to do something afterwards, then the 'h' extension
> or hangup handlers are used to execute logic when the channel is hung up.

Okay, sounds simple enough - thanks,


Antony.

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Re: [asterisk-users] VoiceMail() stop dialplan processing

[Joshua C. Colp]

On Mon, Oct 3, 2022 at 9:11 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> Hi.
>
> I have a dialplan which calls the VoiceMail() application, and I'm getting
> the
> following behaviour:
>
>  - if the inbound caller leaves a message, then presses #, and then
> presses 1
> to accept the recording, everything works as expected and the dialplan
> continues processing after the line containing VoiceMail()
>
>  - if the inbound caller leaves a message and then hangs up, the diaplan
> simply stops executing with a message such as:
>
> [2022-10-03 13:02:23.355976] pbx VERBOSE[19022][C-0556]: pbx.c:4413 in
> __ast_pbx_run: Spawn extension (RecordVM, 00xx74xx88xx90, 2) exited
> non-zero
> on 'SIP/TrunkOne-0c12'
>
> The subsequent commands in the dialplan do not get processed.
>

This is fundamentally how dialplan works. If a channel hangs up, then
normal dialplan execution stops.


>
>
> Can anyone suggest either why this would happen and how to get the
> dialplan to
> continue processing under all circumstances, or at least how to
> investigate
> futher what is causing this to happen?
>
> I'm sure that leaving a message and hanging up the call should be valid
> because that's what the default greeting message tells people they can do.
>

It is. If you're needing to do something afterwards, then the 'h' extension
or hangup handlers are used to execute logic when the channel is hung up.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] voicemail message not deleted

[Michael Keuter]

> Am 26.07.2021 um 07:28 schrieb Fourhundred Thecat <400the...@gmx.ch>:
> 
> Hello,
> 
> I have this in my voicemail.conf:
> 
>  attach=yes
> 
>  delete=yes
> 
> I do get an email when new voicemail is received, and I do get the
> voicemail message as attachment.
> 
> However, the original message is not deleted from the sevber.
> 
> How do I delete the message, after it has been sent per email as
> attachment? I don't want to store messages on the server indefinitely.
> 
> thanks,
> 
> -- 

I think you need to set "delete=yes" as option per mailbox account. 

100 => 1234,Test,,,delete=yes

The global setting is only an example.

Michael

http://www.mksolutions.info




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Re: [asterisk-users] Voicemail help when listening to messages

[Nathan Ward]

Hi,

I have replicated this on a fresh 15.6.0 install, chan_sip, voicemail plain, 
vanilla config (whatever ships with the tucny.com RPMs) other than creating a 
SIP extension and voicemail user.

Should I file a bug?

> On 4/09/2018, at 3:46 PM, Nathan Ward  wrote:
> 
> Hi,
> 
> I have a weird issue, unsure if it’s due to a bug, or configuration on my 
> end. We’re on 14.7.7. I’ve looked at the app_voicemail.c code, and see no 
> changes in this area of the code until the current version so don’t think age 
> of the code is an issue here (but happy to be proven wrong!).
> 
> When hitting * for help when listening to messages (dial in, press 1, then 
> *), we get the following options:
> 
> press 1 for new messages
> press 2 to change folders
> press 3 for advanced options
> press 0 for mailbox options
> press 1 to listen to new messages *
> press 2 to access messages saved in other folders *
> press 3 to record a message for another mailbox *
> press 0 for greetings and password management *
> press 5 to repeat the current message
> press 6 to play the next message
> press 7 to delete this message
> press 8 to forward the message to another user
> press 9 to save this message
> press * for help or # to exit
> 
> I’ve marked 4 items with *s, these are items which seem out of place, as the 
> other options largely work.
> 
> I’ve been poking around, and it seems to be that this is happening because 
> `skipadvanced` is set to 1, though, it this is set to 0 it would (I think) be:
> press 1 for new messages
> press 2 to change folders
> press 3 for advanced options
> press 0 for mailbox options
> press 3 for advanced options *
> press 5 to repeat the current message
> press 6 to play the next message
> press 7 to delete this message
> press 8 to forward the message to another user
> press 9 to save this message
> press * for help or # to exit
> 
> Note the item with a * where advanced options is offered again (once from 
> vm-opts then later from vm-advopts).
> 
> I am not clear on how this is supposed to work, as both with and without 
> skipadvanced set seems weird. Can anyone help?
> 
> --
> Nathan Ward
> 


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Re: [asterisk-users] Voicemail: search for name in a phonebook

[Luca Bertoncello]

John Kiniston  schrieb:

> Yes, You could do easily this either with the internal asterisk database or
> with something like func_odbc as a source for the data.
> 
> In the context you receive your incoming calls you do a lookup against one
> of the above data sources using the CALLERID(NUM) and change CALLERID(NAME)
> to be the name you set.

Thanks a lot!

I found this page:

http://deepliquid.com/blog/archives/59

and I successfully got it working!

Regards
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voicemail: search for name in a phonebook

[John Kiniston]

Yes, You could do easily this either with the internal asterisk database or
with something like func_odbc as a source for the data.

In the context you receive your incoming calls you do a lookup against one
of the above data sources using the CALLERID(NUM) and change CALLERID(NAME)
to be the name you set.

On Wed, Sep 20, 2017 at 1:04 PM, Luca Bertoncello 
wrote:

> Hi list!
>
> I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly.
> I configured a voicemail and I receive an E-Mail with some information
> about
> the call.
> Again, wonderful!
>
> Now my wish: I'd like to have Asterisk to search the caller in a list file
> and send me the name corresponding to the number in the E-Mail of
> voicemail.
> Is it possible?
>
> I currently use ${VM_CALLERID} in emailbody and it gives, of course, the
> phone number...
>
> Thanks a lot!
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] Voicemail asking for login

[James Cloos]

> "DC" == D'Arcy Cain  writes:

DC> I did debug 10 and saved the console output into files which I
DC> compared side by side.  No material difference.

In that case I'd add more debug statements to apps/app_voicemail.c (in
vm_exec()), including a log at the start of what is in *data and args.

Looking at it, it only plays vm-whichbox when ast_strlen_zero(data),
which implies that the args to Voicemail are not making it through.

-JimC
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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 04:07 PM, James Cloos wrote:

I enable full log and run 'core set debug 9' before doing a pair of
tests.

(The full log is easier to grep than the console output.)

Then compare a working vs stocktrans2 side by side.


I did debug 10 and saved the console output into files which I compared 
side by side.  No material difference.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[James Cloos]

I enable full log and run 'core set debug 9' before doing a pair of
tests.

(The full log is easier to grep than the console output.)

Then compare a working vs stocktrans2 side by side.

-JimC
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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 12:52 PM, J Montoya wrote:

On Thursday 20 Apr 2017, D'Arcy Cain wrote:

On 2017-04-20 12:23 PM, D'Arcy Cain wrote:

Here is the full dialplan for stocktrans2.


I reduced this to the following and I still have the error.

exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
 same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
 same => n,Verbose(0,${CALLERID(all)} going into voice mail for
stocktrans2)
 same => n,VoiceMail(stocktrans2@VoiceMail,u)
 same => n,Hangup()


O.K., so -- assuming that extension "darcy" behaves properly if you simplify
it similarly -- nothing before there can be causing the problem.


Actually, it also failed too when I reduced it.  Turns out that that's 
why I needed to set "_ACCOUNT".  Here is the actual reduced version for 
stocktrans2.  It still fails but substituting "darcy" for "stocktrans2 
works OK.


exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for 
stocktrans2)

same => n,Set(_ACCOUNT=stocktrans2)
same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,Hangup()


What is in your [VoiceMail] context?  Are "stocktrans2" and "darcy" separate
extensions, or is there a catch-all?  What is in the "a" extension  (which
gets called when the * key is pressed) ?


Nothing in [VoiceMail] context except the mailboxes but this precedes 
the context:


[general]
attach=yes
maxsilence=10
maxlogins=3
serveremail=n...@vex.net
format=wav49
fromstring=Vybe Networks Voice Mail
nextaftercmd=yes
forcename=yes
pollmailboxes=yes
pollfreq=5

emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were 
just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox 
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to 
check it when you get a chance.  Thanks!\n\n\t\t\t\t-- Vybe Networks\n


They are separate extensions.  I do not do catch-alls if I can help it. 
Since I generate the configs it is no big deal to expand everything in 
each extension.


; voice mail
  exten => a,1,Verbose(${ACCOUNT} entering mailbox)
same => n,Set(CDR(userfield)=${ACCOUNT})
same => n,VoicemailMain(${ACCOUNT}@VoiceMail)
same => n,Hangup

The VoiceMail extension, as are all the extensions, is in the 
[LocalSets] context.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[J Montoya]

On Thursday 20 Apr 2017, D'Arcy Cain wrote:
> On 2017-04-20 12:23 PM, D'Arcy Cain wrote:
> > Here is the full dialplan for stocktrans2.
> 
> I reduced this to the following and I still have the error.
> 
> exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
>  same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
>  same => n,Verbose(0,${CALLERID(all)} going into voice mail for
> stocktrans2)
>  same => n,VoiceMail(stocktrans2@VoiceMail,u)
>  same => n,Hangup()

O.K., so -- assuming that extension "darcy" behaves properly if you simplify 
it similarly -- nothing before there can be causing the problem.

What is in your [VoiceMail] context?  Are "stocktrans2" and "darcy" separate 
extensions, or is there a catch-all?  What is in the "a" extension  (which 
gets called when the * key is pressed) ?

-- 
JM

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 12:23 PM, D'Arcy Cain wrote:


Here is the full dialplan for stocktrans2.


I reduced this to the following and I still have the error.

exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for 
stocktrans2)

same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,Hangup()

--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote:

This is just screaming "configuration mismatch" -- or, possibly, "latent bug
whereby things parsed in separate places should be treated the same, but are
actually getting treated differently".


I really don't want to be the "my system isn't working so there must be 
a bug in Asterisk" guy but I am certainly starting to suspect it.



I think we are going to need to see your dialplan logic, and maybe your
voicemail.conf, in order to work out what is different between this one user
and all the others.  You might even need to use `hd` to examine the files, just
in case there is a stray non-printing character spoiling things.


Non-printing characters show up in vi so I would have seen that by now.

Every extension is build by a script that takes information from the 
database and does substitutions which is what makes this so baffling. 
Every extension is built exactly the same way.  Here are two voicemail 
entries, the failing one and mine that works.  The only sanitation I did 
was for the password.


stocktrans2 => ,Angelica Douglas,stocktra...@vex.net
darcy => ,Vybe Networks - D'Arcy,da...@vex.net

Here is the full dialplan for stocktrans2.

exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n,Goto(DialCell)
same => n,GotoIf($["x" = "x"]?DialAlt)

same => n(DialAll),Verbose(0,${CALLERID(all)} Calling ${EXTEN} and ALL)
same => n,Dial(SIP/stocktrans2&SIP/thinktel/&SIP/907084,30)
same => n,Goto(VoiceMail)

same => n(DialAlt),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and 
SoftPhone)

same => n,Dial(SIP/stocktrans2&SIP/907084,30)
same => n,Goto(VoiceMail)

same => n(DialCell),GotoIf($["x" = "x"]?DialDesk)
same => n,GotoIf($["${CALLERID(ani)}" = ""]?DialDesk)
same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell "")
same => n,Dial(SIP/stocktrans2&SIP/thinktel/,30)
same => n,Goto(VoiceMail)

same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/stocktrans2,30)

same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for 
stocktrans2)

same => n,Set(_ACCOUNT=stocktrans2)
same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,Hangup()

Here is mine.

exten => darcy,1,Verbose(0,Entering extension darcy)
same => n,GotoIf($["${DEVICE_STATE(SIP/901001)}" = 
"UNAVAILABLE"]?DialCell)

same => n,GotoIf($["x4168035991" = "x"]?DialAlt)

same => n(DialAll),Verbose(0,${CALLERID(all)} Calling ${EXTEN} and ALL)
same => n,Dial(SIP/darcy&SIP/thinktel/4168035991&SIP/901001,30)
same => n,Goto(VoiceMail)

same => n(DialAlt),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and 
SoftPhone)

same => n,Dial(SIP/darcy&SIP/901001,30)
same => n,Goto(VoiceMail)

same => n(DialCell),GotoIf($["x4168035991" = "x"]?DialDesk)
same => n,GotoIf($["${CALLERID(ani)}" = "4168035991"]?DialDesk)
same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell 
"4168035991")

same => n,Dial(SIP/darcy&SIP/thinktel/4168035991,30)
same => n,Goto(VoiceMail)

same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/darcy,30)

same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for darcy)
same => n,Set(_ACCOUNT=darcy)
same => n,VoiceMail(darcy@VoiceMail,u)
same => n,Hangup()

There are some minor differences based on whether they have set up an 
alternate phone but either way it gets to the line that set the CDR 
userfield.  They both ultimately hit one or the other of these lines.


same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,VoiceMail(darcy@VoiceMail,u)

The only "Set" command that might change the environment is the setting 
of "_ACCOUNT" which they both do.  Not sure why I even do that.  Perhaps 
I was planning some other feature that I never finished.  Or does 
VoiceMail() use it?


Someone (in private email for some reason so I won't give his name) 
suggested DumpChan() but I can't seem to make that work, even if I load 
app_dumpchan.


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D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[J Montoya or A J Stiles]

On Wednesday 19 Apr 2017, D'Arcy Cain wrote:
> Yes and  [using something like "1571"]  works just fine for us.  The problem
> is that we are trying
> to deal with the situation where someone calls themselves from another
> phone (internal or external) to pick up their messages.  In every other
> case it asks for their password (which is always numeric) and goes into
> the VM.  This one extension asks for a mailbox.

This is just screaming "configuration mismatch" -- or, possibly, "latent bug 
whereby things parsed in separate places should be treated the same, but are 
actually getting treated differently".

I think we are going to need to see your dialplan logic, and maybe your 
voicemail.conf, in order to work out what is different between this one user 
and all the others.  You might even need to use `hd` to examine the files, just 
in case there is a stray non-printing character spoiling things.

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Note:  Originating address only accepts e-mail from list!  If replying off-
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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-19 11:57 AM, J Montoya or A J Stiles wrote:

I fished this out of an old extensions.conf from a defunct project.  It might
be relevant to your use case:

exten => 1571,1,NoOp(Call to 1571: voicemail retrieval)
exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)})
exten => 1571,n,NoOp(CLID is ${clid})
exten => 1571,n,VoiceMailMain(${clid},s)


I do something similar using *98.


The upshot of this was, if you dialled 1571 from your own phone, then you got
put straight through to your own voicemail, without logging in.


Yes and that works just fine for us.  The problem is that we are trying 
to deal with the situation where someone calls themselves from another 
phone (internal or external) to pick up their messages.  In every other 
case it asks for their password (which is always numeric) and goes into 
the VM.  This one extension asks for a mailbox.


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Vybe Networks Inc.
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IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[J Montoya or A J Stiles]

On Wednesday 19 Apr 2017, D'Arcy Cain wrote:
> On 2017-04-19 02:39 AM, Pete Mundy wrote:
> > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail
> > 
>  > show users' I can't see why the vm_authenticate function is
>  > failing to read the username :(
> 
> I can answer that one.  It's because we can't enter 'stocktrans2' from a
> telephone so we just hang up.  The question is, why does it ask for the
> mailbox in the first place>

I fished this out of an old extensions.conf from a defunct project.  It might 
be relevant to your use case:

exten => 1571,1,NoOp(Call to 1571: voicemail retrieval)
exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)})
exten => 1571,n,NoOp(CLID is ${clid})
exten => 1571,n,VoiceMailMain(${clid},s)

The AGI script  `lookup_caller_id.agi`  sets the variable  ${clid}  to the 
caller's extension number, after which their mailbox is named  (although there 
is no reason not to set another variable, such as ${mbox} to hold the mailbox 
if you want).  In the call to voicemailmail() we specify this mailbox, and 
also use the  `s`  option to skip password checking  (it was safe in this 
situation to assume that nobody had physical access to a phone who definitely 
should not have had access to its user's voicemail messages).

The upshot of this was, if you dialled 1571 from your own phone, then you got 
put straight through to your own voicemail, without logging in.  

-- 
JM or AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-19 02:39 AM, Pete Mundy wrote:

Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail

> show users' I can't see why the vm_authenticate function is
> failing to read the username :(

I can answer that one.  It's because we can't enter 'stocktrans2' from a 
telephone so we just hang up.  The question is, why does it ask for the 
mailbox in the first place>



If I were any good at coding in C, I'd probably look inside

> app_voicemail.c around line number 10671 and see if I could
> determine how it reads the username and maybe throw some hacky
? debug output in there to try and determine at which point of
> that process it's failing. But I'm no good at coding in that
> language, so will have to defer to others to help.

I guess that's my next stop.  Luckily I have kept my C skills somewhat 
active as the chief maintainer for PyGreSQL.  :-)


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Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

> On 19/04/2017, at 4:25 pm, D'Arcy Cain  wrote:
> 
>> Does this mailbox exist?
> 
> Yes.

Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail show users' I 
can't see why the vm_authenticate function is failing to read the username :(

If I were any good at coding in C, I'd probably look inside app_voicemail.c 
around line number 10671 and see if I could determine how it reads the username 
and maybe throw some hacky debug output in there to try and determine at which 
point of that process it's failing. But I'm no good at coding in that language, 
so will have to defer to others to help.

Good work on sending through the console clipping and relevant info. Sorry I 
couldn't resolve it for you.

Anyone else got any other ideas?

Pete




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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-18 08:17 PM, Pete Mundy wrote:

On 19/04/2017, at 7:58 am, D'Arcy Cain mailto:da...@vybenetworks.com>> wrote:


Everything looks the same as another one that works except for two
things.  The one that works doesn't have the "Probation passed" lines.
I am not sure if that is even part of this call.  The other is the
line with "Playing 'vm-login.gsm'" in it.  at that point the working
one has this:



Presumably also the line containing 'vm_authenticate: Couldn't read
username' also doesn't appear in the output on a working mailbox either?


Exactly.  Since it is not all digits it can't be entered.


I think that's the place to concentrate your efforts.

It shows shortly after the attempt by VoiceMailMain to enter mailbox
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?


Yes.


Can you show the equivalent line from a working mailbox (so we can see
if it also uses the context 'VoiceMail', or maybe something else
instead, like 'default'?).


"" <6477190146> going into voice mail for alex<<<
-- Executing [alex@LocalSets:19] Set("SIP/thinktel-0181", 
"_ACCOUNT=alex") in new stack<<<
-- Executing [alex@LocalSets:20] VoiceMail("SIP/thinktel-0181", 
"alex@VoiceMail,u") in new stack<<<
--  Playing 
'/var/spool/asterisk/voicemail/VoiceMail/alex/unavail.gsm' (language 
'en')<<<
[Apr 18 11:56:47] DTMF[-1][C-0004c485]: channel.c:4215 __ast_read: DTMF 
begin '*' received on SIP/thinktel-0181<<<
[Apr 18 11:56:47] DTMF[-1][C-0004c485]: channel.c:4219 __ast_read: DTMF 
begin ignored '*' on SIP/thinktel-0181<<<
[Apr 18 11:56:48] DTMF[-1][C-0004c485]: channel.c:4129 __ast_read: DTMF 
end '*' received on SIP/thinktel-0181, duration 280 ms<<<
[Apr 18 11:56:48] DTMF[-1][C-0004c485]: channel.c:4199 __ast_read: DTMF 
end passthrough '*' on SIP/thinktel-0181<<<
-- Executing [a@LocalSets:1] Verbose("SIP/thinktel-0181", "alex 
entering mailbox") in new stack<<<

alex entering mailbox<<<
-- Executing [a@LocalSets:2] Set("SIP/thinktel-0181", 
"CDR(userfield)=alex") in new stack<<<
-- Executing [a@LocalSets:3] VoiceMailMain("SIP/thinktel-0181", 
"alex@VoiceMail") in new stack<<<

--  Playing 'vm-password.gsm' (language 'en')<<<
[Apr 18 11:56:53] WARNING[-1][C-0004c485]: app_voicemail.c:10671 
vm_authenticate: Unable to read password<<<


I hung up before entering the password but it does work when the user 
does it.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-18 08:31 PM, Victor Villarreal wrote:

Maybe excecuting the following command at Asterisk console, will help you:

asterisk> voicemail show users

And you will get a list of all mailbox configured in your system. Search
for the user with problems.


VoiceMail  stocktrans2 Angelica Douglas 12

Definitely there.  In fact, I generate all the configs from a database 
with a script so I would be very surprised if one user was different 
from another.


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Re: [asterisk-users] Voicemail asking for login

[Victor Villarreal]

Hi Darcy,

What Pete think is correct.

Maybe excecuting the following command at Asterisk console, will help you:

asterisk> voicemail show users

And you will get a list of all mailbox configured in your system. Search
for the user with problems.

Finally, in the Asterisk wiki you can find more info:

https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes

Cheers

El 18 abr. 2017 21:18, "Pete Mundy"  escribió:

On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:



Everything looks the same as another one that works except for two things.
The one that works doesn't have the "Probation passed" lines. I am not sure
if that is even part of this call.  The other is the line with "Playing
'vm-login.gsm'" in it.  at that point the working one has this:




Presumably also the line containing 'vm_authenticate: Couldn't read
username' also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.
It shows shortly after the attempt by VoiceMailMain to enter mailbox
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if
it also uses the context 'VoiceMail', or maybe something else instead, like
'default'?).

Pete


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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

> On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:
> 
> 
> Everything looks the same as another one that works except for two things.  
> The one that works doesn't have the "Probation passed" lines. I am not sure 
> if that is even part of this call.  The other is the line with "Playing 
> 'vm-login.gsm'" in it.  at that point the working one has this:
> 

Presumably also the line containing 'vm_authenticate: Couldn't read username' 
also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.

It shows shortly after the attempt by VoiceMailMain to enter mailbox 
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if it 
also uses the context 'VoiceMail', or maybe something else instead, like 
'default'?).

Pete



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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-18 02:42 AM, Pete Mundy wrote:

Try this:

asterisk -r
core set verbose 10
[get user to trigger fault]
[examine console output, and post to list if still unclear]

If you don't solve it yourself, then we'll be able to help further once
we've seen the output.


I can't see much more than at my previous debug level but here it is 
anyway.  Due to line wrapping I added "<<<" to the end of each line in 
case it is not clear where the actual line endings are.


"Alex Chernyshev" <4164251212> going into voice mail for stocktrans2<<<
-- Executing [stocktrans2@LocalSets:19] Set("SIP/alex-0175", 
"_ACCOUNT=stocktrans2") in new stack<<<
-- Executing [stocktrans2@LocalSets:20] 
VoiceMail("SIP/alex-0175", "stocktrans2@VoiceMail,u") in new stack<<<
   > 0x7f7fea5dc000 -- Probation passed - setting RTP source 
address to 72.143.94.110:28503<<<
--  Playing 
'/var/spool/asterisk/voicemail/VoiceMail/stocktrans2/unavail.gsm' 
(language 'en')<<<
   > 0x7f7fea5dc000 -- Probation passed - setting RTP source 
address to 72.143.94.110:28503<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4215 __ast_read: DTMF 
begin '*' received on SIP/alex-0175<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4219 __ast_read: DTMF 
begin ignored '*' on SIP/alex-0175<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4129 __ast_read: DTMF 
end '*' received on SIP/alex-0175, duration 160 ms<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4199 __ast_read: DTMF 
end passthrough '*' on SIP/alex-0175<<<
-- Executing [a@LocalSets:1] Verbose("SIP/alex-0175", 
"stocktrans2 entering mailbox") in new stack<<<

stocktrans2 entering mailbox<<<
-- Executing [a@LocalSets:2] Set("SIP/alex-0175", 
"CDR(userfield)=stocktrans2") in new stack<<<
-- Executing [a@LocalSets:3] VoiceMailMain("SIP/alex-0175", 
"stocktrans2@VoiceMail") in new stack<<<

--  Playing 'vm-login.gsm' (language 'en')<<<
[Apr 18 11:45:49] WARNING[-1][C-0004c47b]: app_voicemail.c:10627 
vm_authenticate: Couldn't read username<<<


Everything looks the same as another one that works except for two 
things.  The one that works doesn't have the "Probation passed" lines. I 
am not sure if that is even part of this call.  The other is the line 
with "Playing 'vm-login.gsm'" in it.  at that point the working one has 
this:


--  Playing 'vm-password.gsm' (language 'en')

Not sure if that's useful information since it just describes the 
original issue - that it asks for a login instead of a password.



--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Voicemail asking for login

[Israel Gottlieb]

 Does he have the same voicemail context?From: p...@fiberphone.co.nzSent: April 18, 2017 9:43 AMTo: asterisk-users@lists.digium.comReply-to: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Voicemail asking for login  Hi D'ArcyOn 18/04/2017, at 5:17 am, D'Arcy Cain <da...@vybenetworks.com> wrote:One user (that we know of so far) has a different experience.  In that case they are asked for a mailbox number first.  I have tried searching for this issue but nothing seems to apply.  Most discussions are about "*97" vs. "*98".  Can anyone suggest another field of enquiry?Try this:	asterisk -r	core set verbose 10	[get user to trigger fault]	[examine console output, and post to list if still unclear]If you don't solve it yourself, then we'll be able to help further once we've seen the output.HTH,Pete-- 
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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

Hi D'Arcy

> On 18/04/2017, at 5:17 am, D'Arcy Cain  wrote:
> 
> 
> One user (that we know of so far) has a different experience.  In that case 
> they are asked for a mailbox number first.  
> 
> I have tried searching for this issue but nothing seems to apply.  Most 
> discussions are about "*97" vs. "*98".  Can anyone suggest another field of 
> enquiry?


Try this:

asterisk -r
core set verbose 10
[get user to trigger fault]
[examine console output, and post to list if still unclear]

If you don't solve it yourself, then we'll be able to help further once we've 
seen the output.

HTH,

Pete




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Re: [asterisk-users] Voicemail notification by email is missing CallerID info

[Tech Support]

I’ll go through it and see what I missed. I can't thank you enough!

John

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H
Sent: Saturday, February 18, 2017 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail notification by email is missing 
CallerID info

 

This is what comes with voicemail.conf.sample - works for me!

; Change the from, body and/or subject, variables:

; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,

; VM_CIDNAME, VM_DATE

; Additionally, on forwarded messages, you have the variables:

; ORIG_VM_CALLERID, ORIG_VM_CIDNUM, ORIG_VM_CIDNAME, ORIG_VM_DATE

; You can select between two variables by using dialplan functions, e.g.

; ${IF(${ISNULL(${ORIG_VM_DATE})}?${VM_DATE}:${ORIG_VM_DATE})}

;

; Note: The emailbody config row can only be up to 512 characters due to a

;   limitation in the Asterisk configuration subsystem.

;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}

; The following definition is very close to the default, but the default shows

; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown

; caller", if they are both null.

;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left 
a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from 
${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a 
chance.  Thanks!\n\n\t\t\t\t--Asterisk\n

;

; Note: ${IF()} strips spacing at the beginning and end of its true and false

; values, so a newline cannot be placed at either location.  The word 'so' is

; therefore duplicated, in order for the newline to be interpreted correctly.

;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just 
${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR} long 
message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on 
${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent 
by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it 
when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n

 

On 18 February 2017 at 16:35, Tech Support  wrote:
> All;
>
> I am running Asterisk 11.6-cert16 and I have voicemail setup so
> voicemail messages are sent as email attachments. That works fine. However,
> the body of the email contains the CallerID(name), but is missing the
> CallerID(num). For example, the email body looks like this:
>
>
>
>   Just wanted to let you know you were just left a 0:21 long message
> (number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17,
> 2017 at 04:48:38 PM so you might want to check it when you get a chance.
> Thanks!
>
>
>
> Checking the CDR’s shows that both the name and number were recorded by
> Asterisk. Am I missing something obvious? Is it a simple config option in
> voicemail.conf? Any insight at all would be greatly appreciated.
>
> Thanks;
>
> John V.
>
>
>
>
> --
> _
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>
> Check out the new Asterisk community forum at:
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>
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>
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Re: [asterisk-users] Voicemail notification by email is missing CallerID info

[Jonathan H]

This is what comes with voicemail.conf.sample - works for me!

; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
; VM_CIDNAME, VM_DATE
; Additionally, on forwarded messages, you have the variables:
; ORIG_VM_CALLERID, ORIG_VM_CIDNUM, ORIG_VM_CIDNAME, ORIG_VM_DATE
; You can select between two variables by using dialplan functions, e.g.
; ${IF(${ISNULL(${ORIG_VM_DATE})}?${VM_DATE}:${ORIG_VM_DATE})}
;
; Note: The emailbody config row can only be up to 512 characters due to a
;   limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default
shows
; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an
unknown
; caller", if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just
left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to
check it when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n
;
; Note: ${IF()} strips spacing at the beginning and end of its true and
false
; values, so a newline cannot be placed at either location.  The word 'so'
is
; therefore duplicated, in order for the newline to be interpreted
correctly.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just
${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR}
long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" =
"${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on
${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance.
Thanks!\n\n\t\t\t\t--Asterisk\n

On 18 February 2017 at 16:35, Tech Support  wrote:
> All;
>
> I am running Asterisk 11.6-cert16 and I have voicemail setup so
> voicemail messages are sent as email attachments. That works fine.
However,
> the body of the email contains the CallerID(name), but is missing the
> CallerID(num). For example, the email body looks like this:
>
>
>
>   Just wanted to let you know you were just left a 0:21 long message
> (number 13) in mailbox 101 from WIRELESS CALLER, on Friday, February 17,
> 2017 at 04:48:38 PM so you might want to check it when you get a chance.
> Thanks!
>
>
>
> Checking the CDR’s shows that both the name and number were recorded by
> Asterisk. Am I missing something obvious? Is it a simple config option in
> voicemail.conf? Any insight at all would be greatly appreciated.
>
> Thanks;
>
> John V.
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] voicemail greeting

[Bertrand LUPART - Linkeo.com]

> hi.i managed to record my voicemail greeting. the only problem is that after 
> my greeting the caller hear '...please leave your message after the tone. 
> when done press the pound key or hangup.' is there a way to get rid of that?
> Ideally i would like to have my own recording and then the beep sound.
> 

Try option s :

https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail

Regards,

-- 
Bertrand LUPART

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Re: [asterisk-users] VoiceMail - Allow * for only some users

[Andrew Ruthven]

Hi John,

Ah ha!  Excellent. That works.

Now for a further tweak, in my stdexten I set voicemail_option with
with b or u, as appropriate and use ${voicemail_option) instead of
option in the call to Voicemail below so the correct prompt is used.

Thank you!

On Thu, 2016-07-21 at 14:53 -0700, John Kiniston wrote:
> I think you almost have it.
> 
> In your vmfwd context have a wildcard match that sends the caller
> back to the originating voicemail and then define specific extensions
> that are allowed to forward.
> 
> 
> [vmfwd]
> exten => _,1,Voicemail(box@context,option)
>  same =>  n,Hangup
> 
> ; Andrew Ruthven
> exten => 7231,1,Set(CALLERID(number)=yyy)
> same => n,Goto(pstn,xxx,1)
> 
> On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven  yst.net.nz> wrote:
> > Hey,
> > 
> > I have free calling to between DDIs and cellphones on our group
> > plan. I
> > figure it'd be nice to allow staff with those cellphones to be able
> > to
> > forward callers to their VoiceMail to their cellphones using the *
> > feature.
> > 
> > I have a standard extension macro that has VoiceMail support.
> > So far I've done this by duplicating the standard extension macro,
> > and
> > adding this rule (where ARG1 is the extension):
> > 
> >   exten => a,1,Goto(vmfwd,${ARG1},1)
> > 
> > Then in the vmfwd context I have rules like this (I need to set the
> > CALLERID(number) so our SIP provider accepts the call):
> > 
> >   ; Andrew Ruthven
> >   exten => 7231,1,Set(CALLERID(number)=yyy)
> >   exten => 7231,n,Goto(pstn,xxx,1)
> > 
> > Which is working nicely. But, I thought, can I simplify this and
> > just
> > have one macro?
> > 
> > So I've tried doing the following to fold it into my standard
> > extension
> > macro:
> > 
> > 1) Tried using a/_7231 but that didn't match (well, it was worth a
> > try)
> > 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my
> > extension,
> > but if I disable the 7231 rules in vmfwd, I get:
> > 
> >   [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
> > __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to
> > invalid
> > extension but no invalid handler:
> > context,exten,priority=vmfwd,7231,1
> > 
> >   and the call hangs up, not a very nice user experience.
> > 
> > The second option could work, as long as the user lands back into
> > VoiceMail if there is no valid extension. I thought about using
> > GoSub,
> > but how do I get the caller back into VoiceMail?
> > 
> > I've done a bunch of searching for this, but haven't found any
> > general
> > solutions. Is it possible to do what I'm trying to achieve, or is
> > there
> > a better approach?
> > 
> > This is Asterisk 11.13.
> > 
> > Cheers,
> > Andrew
> > 
> > --
> > 
> > Andrew Ruthven, Wellington, New Zealand
> > MIITP, CITPNZ
> > 
> > At work: andrew.ruth...@catalyst.net.nz
> > At home: and...@etc.gen.nz
> > Card   : http://qr.catalyst.net.nz/907675e1
> > Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
> > GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
> > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
> > 
> > 
> > 
> > 
> > 
> > --
> > ___
> > __
> > -- 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
> 
> 
> 
-- 

Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ

At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card   : http://qr.catalyst.net.nz/907675e1
Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org





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Re: [asterisk-users] VoiceMail - Allow * for only some users

[John Kiniston]

I think you almost have it.

In your vmfwd context have a wildcard match that sends the caller back to
the originating voicemail and then define specific extensions that are
allowed to forward.


[vmfwd]
exten => _,1,Voicemail(box@context,option)
 same =>  n,Hangup

; Andrew Ruthven
exten => 7231,1,Set(CALLERID(number)=yyy)
same => n,Goto(pstn,xxx,1)

On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven <
andrew.ruth...@catalyst.net.nz> wrote:

> Hey,
>
> I have free calling to between DDIs and cellphones on our group plan. I
> figure it'd be nice to allow staff with those cellphones to be able to
> forward callers to their VoiceMail to their cellphones using the *
> feature.
>
> I have a standard extension macro that has VoiceMail support.
> So far I've done this by duplicating the standard extension macro, and
> adding this rule (where ARG1 is the extension):
>
>   exten => a,1,Goto(vmfwd,${ARG1},1)
>
> Then in the vmfwd context I have rules like this (I need to set the
> CALLERID(number) so our SIP provider accepts the call):
>
>   ; Andrew Ruthven
>   exten => 7231,1,Set(CALLERID(number)=yyy)
>   exten => 7231,n,Goto(pstn,xxx,1)
>
> Which is working nicely. But, I thought, can I simplify this and just
> have one macro?
>
> So I've tried doing the following to fold it into my standard extension
> macro:
>
> 1) Tried using a/_7231 but that didn't match (well, it was worth a try)
> 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension,
> but if I disable the 7231 rules in vmfwd, I get:
>
>   [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
> __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid
> extension but no invalid handler: context,exten,priority=vmfwd,7231,1
>
>   and the call hangs up, not a very nice user experience.
>
> The second option could work, as long as the user lands back into
> VoiceMail if there is no valid extension. I thought about using GoSub,
> but how do I get the caller back into VoiceMail?
>
> I've done a bunch of searching for this, but haven't found any general
> solutions. Is it possible to do what I'm trying to achieve, or is there
> a better approach?
>
> This is Asterisk 11.13.
>
> Cheers,
> Andrew
>
> --
>
> Andrew Ruthven, Wellington, New Zealand
> MIITP, CITPNZ
>
> At work: andrew.ruth...@catalyst.net.nz
> At home: and...@etc.gen.nz
> Card   : http://qr.catalyst.net.nz/907675e1
> Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
> GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
> LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
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Re: [asterisk-users] VoiceMail and SMS

[A J Stiles]

On Friday 15 Jul 2016, Joaquin Alzola wrote:
> Hi Guys
> 
> I am asking too many questions because we would like to use Asterisk first
> as a proof of Concept and check from there were it goes.
> 
> - Does the Voicemail have the option of SMS notification on new drop
> messages (we have an SMSC so we will use that one). 

Asterisk Voicemail can certainly send an e-mail when a message is left.  By 
cunning use of a procmail recipe, this can be used to send an SMS or do 
anything else.

> - What is the best Linux OS to install Asterisk in?

The one with which you are most familiar.

> - What throughput does it stand 1 machine with about 8GB Ram and 4 CPUs? We
> plan to add  couple but just checking for a single one.

I've seen boxes with 2 cores, 4 GB RAM, 8 outside lines, all calls recorded 
using MixMonitor and no swapping; 4 cores, 8 GB RAM, 20 outside lines on an 
ISDN30 and more via SIP trunks, MixMonitor recording and again no swapping.

> - Does it hava a max capacity?

Probably, but good luck trying to find it  :)  


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] VoiceMail and SMS

[Tim S]

How many users are you thinking of supporting?  For a large-scale setup you
might want to take a look at Kamailio as a front-end - if you even think
you're going to get a high user volume you may want to start out with a
Kamailio front-end so that you don't have to start over from scratch when
it outgrows an Asterisk only setup.

I prefer Debian/Ubuntu over RedHat/CentOS - but if you're thinking of using
this in a company environment, I'd recommend engaging with your IT people
to find out what THEIR preference is.

Planning capacity with the information you've provided is difficult - is
the network card 10Base-T, Gigabit, 100-gigabit?  Do you only have 56kbps
dial-up service to the server, or a full 10Gbps internet connection at a
carrier-neutral colocation datacenter on a fiber backbone?  Are the CPU
cores 15-year-old Pentium or a current Broadwell-E?  Are the CPU cores real
or on a massively over-provisioned VM host?  Do you have to do a bunch of
transcoding inbound and outbound?  Is the machine doing anything other than
voicemail?  In general, the maximum capacity is the point just before when
the quality begins to drop (about 90-95% total system load).  Not very
scientific I know, but the answer is extremely
hardware/infrastructure/setup dependent.

I haven't personally played with any of Asterisk's internal SMS
functionality, but I have been meaning to.  Since the earlier days I've
relied on the email functions to handle interfacing notifications.
Asterisk can send emails as a notification, so I configured the default
email to a notification handler, which would do a speech recognition on the
voicemail file, send an SMS using an SMSC (Nexmo in my case), and then send
an email to the user with a text transcript of the voicemail as well as the
audio file as an attachment.  I'm sure there's a better way now, I coded
this up a while ago.  My way is probably not the "right way", but like many
things with computers there is the way that works today, the way that works
better tomorrow, and eventually the best practice way that emerges after a
few years.  Gotta keep maintaining your work.

-Tim


On Fri, Jul 15, 2016 at 8:29 AM, Joaquin Alzola 
wrote:

> Hi Guys
>
> I am asking too many questions because we would like to use Asterisk first
> as a proof of Concept and check from there were it goes.
>
> - Does the Voicemail have the option of SMS notification on new drop
> messages (we have an SMSC so we will use that one).
> - What is the best Linux OS to install Asterisk in?
> - What throughput does it stand 1 machine with about 8GB Ram and 4 CPUs?
> We plan to add  couple but just checking for a single one.
> - Does it have a max capacity?
>
> Thanks for your time.
>
> BR
>
> Joaquin
> This email is confidential and may be subject to privilege. If you are not
> the intended recipient, please do not copy or disclose its content but
> contact the sender immediately upon receipt.
>
> --
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] VoiceMail Audio playing

[Joaquin Alzola]

> No.  The VoiceMail server takes care of all that itself; it delivers the 
> broadcast and records the messages.

Thanks AJ.
This email is confidential and may be subject to privilege. If you are not the 
intended recipient, please do not copy or disclose its content but contact the 
sender immediately upon receipt.
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Re: [asterisk-users] VoiceMail Audio playing

[A J Stiles]

On Friday 15 Jul 2016, Joaquin Alzola wrote:
> Hi Madushan
> 
> Maybe I was not clear …. After SIP negotiation and SDP set up on the
> VoiceMail Server ….
> 
> Is there  a file to specify a MGw (the machine that deliver RTP packages to
> end user)?

No.  The VoiceMail server takes care of all that itself; it delivers the 
broadcast and records the messages.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] VoiceMail Audio playing

[Joaquin Alzola]

> Asterisk does not separate things like this. For media originating from it 
> the source will always be it. That is if you do a SIP call to Asterisk then 
> media will come from that same Asterisk.

Joshua ok perfect so Asterisk already have the play module incorporated.
That’s great to hear so no need to integrate it to a MediaGatwey or SBC.
This email is confidential and may be subject to privilege. If you are not the 
intended recipient, please do not copy or disclose its content but contact the 
sender immediately upon receipt.
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Re: [asterisk-users] VoiceMail Audio playing

[Joshua Colp]

Joaquin Alzola wrote:

Hi Madushan

Maybe I was not clear …. After SIP negotiation and SDP set up on the
VoiceMail Server ….

Is there a file to specify a MGw (the machine that deliver RTP packages
to end user)?


Asterisk does not separate things like this. For media originating from 
it the source will always be it. That is if you do a SIP call to 
Asterisk then media will come from that same Asterisk.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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Re: [asterisk-users] VoiceMail Audio playing

[Joaquin Alzola]

Hi Madushan

Maybe I was not clear …. After SIP negotiation and SDP set up on the VoiceMail 
Server ….

Is there  a file to specify a MGw (the machine that deliver RTP packages to end 
user)?

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Madushan Geethanga
Sent: 15 July 2016 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] VoiceMail Audio playing

Hi,
VoiceMailMain is used to retrieve voice mails

http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain
Best Regards,
Madushan

On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzola 
mailto:joaquin.alz...@lebara.com>> wrote:
Hi Guys

Which module on Asterisk is the one in charge of playing the VoiceMail Server 
Audio to the end customer?
I have work with MRFP but is it a module included in the SW? Need and external 
source?

BR

Joaquin
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Re: [asterisk-users] VoiceMail Audio playing

[Madushan Geethanga]

Hi,

VoiceMailMain is used to retrieve voice mails

http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain

Best Regards,
Madushan

On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzola 
wrote:

> Hi Guys
>
>
>
> Which module on Asterisk is the one in charge of playing the VoiceMail
> Server Audio to the end customer?
>
> I have work with MRFP but is it a module included in the SW? Need and
> external source?
>
>
>
> BR
>
>
>
> Joaquin
> This email is confidential and may be subject to privilege. If you are not
> the intended recipient, please do not copy or disclose its content but
> contact the sender immediately upon receipt.
>
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Re: [asterisk-users] voicemail: duration while leaving a message

[Mamadou NGOM]

Thank you joshua.Le 9 mai 2016 à 16:00, Joshua Colp  a écrit :Mamadou NGOM wrote:Hello list,Kia ora,I am asking when a caller want to leave a message to a mailbox with theapplication voicemailHow i can limit the duration for exemple 30 seconds.exten => _X,n,VoiceMail(${Caller_number},s)Is there a option which allows me to do it, somebody to help me.This can be configured in voicemail.conf using the "maxsecs" configuration option. I don't believe this is exposed using the Voicemail application options, just using the config file.Cheers,-- Joshua ColpDigium, Inc. | Senior Software Developer445 Jan Davis Drive NW - Huntsville, AL 35806 - USCheck us out at: www.digium.com & www.asterisk.org-- _-- 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/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersMamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06 72 45 23 03Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 

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Re: [asterisk-users] voicemail: duration while leaving a message

[Joshua Colp]

Mamadou NGOM wrote:

Hello list,


Kia ora,


I am asking when a caller want to leave a message to a mailbox with the
application voicemail

How i can limit the duration for exemple 30 seconds.

exten => _X,n,VoiceMail(${Caller_number},s)

Is there a option which allows me to do it, somebody to help me.


This can be configured in voicemail.conf using the "maxsecs" 
configuration option. I don't believe this is exposed using the 
Voicemail application options, just using the config file.


Cheers,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Voicemail using object storage?

[Andrew Ruthven]

I'd say using s3fs (or similar) is an approach, but if VoiceMail had
support baked into it for S3, then the integration would be better.

I'll look into using one the FUSE based approaches as a stop-gap
measure. ;)

On Tue, 2016-02-16 at 13:12 +0100, Olivier wrote:
> Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be
> mounted on Linux and thus allow any application like Asterisk make
> use of it ?
> 
> [1] https://github.com/s3fs-fuse/s3fs-fuse
> 
> 2016-02-16 1:05 GMT+01:00 Andrew Ruthven  .nz>:
> > Hey,
> > 
> > I've found a bit of chatter about people using hacks to copy
> > voicemail
> > messages into object storage (like S3) after they've been recorded.
> > But
> > I was wondering if any work has been done on the VoiceMail app to
> > actually store and retrieve messages to/from an object store?
> > 
> > Cheers,
> > Andrew
> > --
> > Andrew Ruthven, Wellington, New Zealand
> > MIITP, ITCP
> > 
> > At work: andrew.ruth...@catalyst.net.nz
> > At home: and...@etc.gen.nz
> > Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
> > GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
> > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
> > 
> > 
> > 
> > --
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> > 
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Re: [asterisk-users] Voicemail using object storage?

[Olivier]

Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be
mounted on Linux and thus allow any application like Asterisk make use of
it ?

[1] https://github.com/s3fs-fuse/s3fs-fuse

2016-02-16 1:05 GMT+01:00 Andrew Ruthven :

> Hey,
>
> I've found a bit of chatter about people using hacks to copy voicemail
> messages into object storage (like S3) after they've been recorded. But
> I was wondering if any work has been done on the VoiceMail app to
> actually store and retrieve messages to/from an object store?
>
> Cheers,
> Andrew
> --
> Andrew Ruthven, Wellington, New Zealand
> MIITP, ITCP
>
> At work: andrew.ruth...@catalyst.net.nz
> At home: and...@etc.gen.nz
> Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
> GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
> LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
>
>
>
> --
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Re: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones

[Bryant Zimmerman]

Richard
  
 Check both the DTMF settings, and the DialPlan string for account 3 on the 
phone.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "Richard Schroeder" 
Sent: Tuesday, February 9, 2016 12:58 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones   
Perhaps this is not limited to Grandstream GXP 2000 phones, but those 
are the phones we are using.
  
 Using FreePBX.
  
 Retrieving a voice message (*97) works fine from Line 1.
 Retrieving a voice message (*98) and picking the extension (Comedian mail) 
works fine from Line 1.
  
 From Line 3, it does not recognize the password. (*97 or *98). The 
extension is installed on Line 3. Retrieving Line 3's voice messages can 
only be done from Line 1 (on any extension on the PBX). Line 3 seems to 
work fine otherwise.
  
 Is this a limitation, or is it some kind of setup issue?
 I can't seem to find anything in the documentation for the phone or 
FreePBX related to this issue.
  
 Anyone? This is frustrating and I will be grateful for any help.
  
 Thank you!
  
 Richard
  
  

--  Richard C. Schroeder
rsch...@gmail.com
rsch...@optonline.net
516-859-1129 - Cell


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Re: [asterisk-users] Voicemail issue on Grandstream GXP2000 phones

[jg]

:

From Line 3, it does not recognize the password.

Did you check whether you have the same DTMF settings for Line 3?

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Re: [asterisk-users] Voicemail: saycid without prefix

[John Kiniston]

Nice!

I didn't know what dialing rules may apply to his location, Your code does
look like an improvement on mine tho.

I love the REGEX function.

Even better, if the first 4 digits are "0049", you could replace them with
> "0"
> as though it was an inland call:
>
> ExecIf(REGEX("^0049."
> ${CALLERID(NUM)})?Set(CALLERID(num)=0${CALLERID(NUM):4}))
>
> --
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Voicemail: saycid without prefix

[A J Stiles]

On Monday 06 Jul 2015, Luca Bertoncello wrote:
> John Kiniston  schrieb:
> > The easiest solution may be to strip the leading zero's off your caller
> > ID before your caller enters the Voicemail app to leave you a message.
> > 
> > 
> > ExecIf(REGEX("^[0][0]."
> > ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
> 
> Thanks!
> 
> I already had this idea and implemented it.
> It works...

Even better, if the first 4 digits are "0049", you could replace them with "0" 
as though it was an inland call:

ExecIf(REGEX("^0049." 
${CALLERID(NUM)})?Set(CALLERID(num)=0${CALLERID(NUM):4}))

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Voicemail: saycid without prefix

[Luca Bertoncello]

John Kiniston  schrieb:

> The easiest solution may be to strip the leading zero's off your caller ID
> before your caller enters the Voicemail app to leave you a message.
> 
> 
> ExecIf(REGEX("^[0][0]."
> ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))

Thanks!

I already had this idea and implemented it.
It works...

Regards
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voicemail: saycid without prefix

[John Kiniston]

The easiest solution may be to strip the leading zero's off your caller ID
before your caller enters the Voicemail app to leave you a message.


ExecIf(REGEX("^[0][0]."
${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))


On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello 
wrote:

> Hi list!
>
> Yesterday I set up a voicemail on my Asterisk 1.8.
> It works as expected, but I'd like to have the CID without unnecessary
> prefix...
>
> Right now, if I call from my mobile phone I hear the complete prefix for my
> mobile number, indeed without "00".
> So I hear "message from 49177...".
>
> How can I set Asterisk to just read the prefix if it's necessary (so that
> calls from german numbers will not have "0049")?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Voicemail ODBC Storage

[Anthony Messina]

On Saturday, October 25, 2014 09:09:57 PM Dan Journo wrote:
> Is there any reason why ODBC voicemail storage requires varchar for most
> fields?  For example, is there anything stopping me using a BIGINT or
> similar for origtime or INT for duration?

It may cause you trouble when using PostgreSQL: 
https://issues.asterisk.org/jira/browse/ASTERISK-24441

-A

-- 
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Voicemail ODBC Storage

[Matthew Jordan]

On Sat, Oct 25, 2014 at 4:09 PM, Dan Journo 
wrote:

> Hi,
>
> Is there any reason why ODBC voicemail storage requires varchar for most
> fields?
> For example, is there anything stopping me using a BIGINT or similar for
> origtime or INT for duration?
>
>
Yes.

app_voicemail uses a message envelope file to hold the metadata regarding
the voice mail. When the ODBC retrieve function pulls the database records,
it writes that data out to a temporary message envelope file for
playback/manipulation by other functions. This process does not examine the
column types, but instead simply looks at the column names and writes the
data values out to the file using the types that it expects each column
name to have.

So, changing those types will not work out well for you.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Voicemail message number off by one when using ODBC storage

[Marie Fischer]

... 'cause message file names start with 0 (msg.wav).

-- 

marie

On 05.10.2014, at 18:45, Leandro Dardini  wrote:

> Hello,
> have you noticed the message num (VM_MSGNUM) is off by one?
> 
> For example, I receive the following message:
> 
> "Just wanted to let you know you were just left a 0:03 long message (number 
> 7)"
> 
> but in attach there is the msg0006.wav
> 
> Leandro
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Re: [asterisk-users] voicemail with odbc

[ProNek]

Hi Rusty,

Noted with thanks.

Regards,
Thet Tun


On Thu, May 29, 2014 at 5:09 PM, Rusty Newton  wrote:

> On Thu, May 29, 2014 at 3:33 AM, ProNek  wrote:
> > Hi,
> >
> > I have some issue with voice mail with ODBC on asterisk 11.7 box. I may
> not
> > understand database functionality on asterisk fully. The most suspected
> area
> > is func_odbc. I already googled but not luck. Your guide is warmly
> welcomed
> >
> 
>
> You already started another mailing list thread on this topic a few
> hours before this. Please don't do that in the future. If you are
> going to post again, just post to the thread you already started
> instead of starting a new one.
>
> Did you double-check your database table carefully against the required
> schema?
> https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage
>
> --
> Rusty Newton
> Digium, Inc. | Community Support Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct: +1 256 428 6200
>
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] voicemail with odbc

[Rusty Newton]

On Thu, May 29, 2014 at 3:33 AM, ProNek  wrote:
> Hi,
>
> I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not
> understand database functionality on asterisk fully. The most suspected area
> is func_odbc. I already googled but not luck. Your guide is warmly welcomed
>


You already started another mailing list thread on this topic a few
hours before this. Please don't do that in the future. If you are
going to post again, just post to the thread you already started
instead of starting a new one.

Did you double-check your database table carefully against the required schema?
https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

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Re: [asterisk-users] Voicemail message to text

[Thorsten Göllner]

Hi,

we implemented ispeech for voice recognition. I works fine. But you have 
to develop an app of your own to do it.


Take a look at http://www.ispeech.org/api (Section 3 Automated Speech 
Recognition).


ispeech let you upload a recorded speex file via http-upload and will 
return the result at once as http-result.


On their website you will find some code also to implement their service 
in any app. It's simple and you will get a quick result.


Best regards
-Thorsten-

Am 20.05.2014 16:35, schrieb Ishfaq Malik:

HI there

I was wondering if anyone has implemented voicemail to text and if so, 
what package is being used to do so?


Thanks in Advance

Ish

--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e:i...@pack-net.co.uk  
w:http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552


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Re: [asterisk-users] Voicemail message to text

[Chris Bagnall]

On 20 May 2014, at 15:35, Ishfaq Malik  wrote:
> I was wondering if anyone has implemented voicemail to text and if so, what 
> package is being used to do so?

With the huge variety of different accents and intonations in human speech 
(even in one country), my experience of all speech-to-text engines has been one 
of poor accuracy at best.

If you need messages-to-text, generally best to use a virtual PA company or 
similar - at least in my experience.

Kind regards,

Chris
-- 
This email is made from 100% recycled electrons


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Re: [asterisk-users] VoiceMail Issue

[Carlos Rojas]

Hi

Could you send us the logs from the asterisk?

Carlos


On Sat, Mar 8, 2014 at 4:03 AM, Phil Daws  wrote:

> Any ideas on why this may not be working please ?
>
> - Original Message -
> From: "Phil Daws" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> Sent: Friday, 28 February, 2014 5:39:54 PM
> Subject: [asterisk-users] VoiceMail Issue
>
> Hello,
>
> am attempting again to resolve an issue with multi-tenancy and the
> forwarding to VMs between mailboxes.  If in a multi-tenancy environment one
> uses custom contexts ie.
>
> [a1-ext1](a1)
> mailbox=101@a1
>
> and the associated voicemail.conf entry:
>
> [a1]
> 101 => 1234,My User 1,ad...@email.com,,tz=eastern|imapuser=ad...@email.com
> |imapfolder=Inbox
> 102 => 1234,My User 2,ad...@email.com,,tz=eastern|imapuser=ad...@email.com
> |imapfolder=Inbox
>
> now if a message is left in mailbox 101 and the user attempts to forward
> the message to mailbox 102 Asterisk responds that mailbox 102 is not found
> in context default!  One can add:
>
> searchcontexts=yes
>
> but that means each mailbox must have a unique number which goes against
> being able to use custom contexts.  I thought by specifying the following
> would fix that:
>
> exten => 7999,1,VoiceMailMain(${CALLERID(num)}@a1) ; Direct mail retrieval
> exten => 7999,n,Hangup()
>
> but it does not.  Have tried many ways to resolve but cannot find a
> resolution.
>
> Any ideas please as would like to get this working ?
>
> Thank you.
>
>
>
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Re: [asterisk-users] VoiceMail Issue

[Phil Daws]

Any ideas on why this may not be working please ?

- Original Message -
From: "Phil Daws" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, 28 February, 2014 5:39:54 PM
Subject: [asterisk-users] VoiceMail Issue

Hello,

am attempting again to resolve an issue with multi-tenancy and the forwarding 
to VMs between mailboxes.  If in a multi-tenancy environment one uses custom 
contexts ie.

[a1-ext1](a1)
mailbox=101@a1

and the associated voicemail.conf entry:

[a1]
101 => 1234,My User 
1,ad...@email.com,,tz=eastern|imapuser=ad...@email.com|imapfolder=Inbox
102 => 1234,My User 
2,ad...@email.com,,tz=eastern|imapuser=ad...@email.com|imapfolder=Inbox

now if a message is left in mailbox 101 and the user attempts to forward the 
message to mailbox 102 Asterisk responds that mailbox 102 is not found in 
context default!  One can add:

searchcontexts=yes

but that means each mailbox must have a unique number which goes against being 
able to use custom contexts.  I thought by specifying the following would fix 
that:

exten => 7999,1,VoiceMailMain(${CALLERID(num)}@a1) ; Direct mail retrieval
exten => 7999,n,Hangup()

but it does not.  Have tried many ways to resolve but cannot find a resolution.

Any ideas please as would like to get this working ?

Thank you.



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Re: [asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

From: "Doug Lytle" 
Sent: Monday, November 25, 2013 6:25 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Voicemail greeting playback issues?

Bryant Zimmerman wrote:
> Hey all
>
> I believe I found the bug in Asterisk 11.xxx If someone can help me
> verify it.

Actually,

I wouldn't consider it a bug.  I've know for years that you need to
answer a channel before you play back audio or strange things can and
will happen.

Doug

-- 
Doug

The real issue here is that issuing an Answer() just before does not seem 
to solve the problem. To work around the issue I have to either put a 
Wait(1) or Dial() some extensions first. It is presenting like if you drop 
into the Voicemail() command too fast during call setup that you have 
issues. This did not occur in 1.8.x. I would be ok if just issuing an 
Answer() would resolve it as this would be normal, but having to slow down 
the dial plan seems off. 

Thanks
Bryant

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Re: [asterisk-users] Voicemail greeting playback issues?

[Matthew Jordan]

On Mon, Nov 25, 2013 at 7:17 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 11/26/2013 12:24 AM, Doug Lytle wrote:
> > Bryant Zimmerman wrote:
> >> Hey all
> >>
> >> I believe I found the bug in Asterisk 11.xxx If someone can help me
> >> verify it.
> >
> > Actually,
> >
> > I wouldn't consider it a bug.  I've know for years that you need to
> > answer a channel before you play back audio or strange things can and
> > will happen.
>
> That's what I do since the 0.x days. IIRC in recent Asterisk versions
> some apps answer before doing anything else. Guess the voicemail app is
> not one of them. I always answer first followed by a small Wait and then
> execute the app.
>
>
VoiceMail does automatically Answer a channel.

I'm going to guess that you have strictrtp enabled (which it is by
default), and that if you cranked up Asterisk verbose logging to at least
4, you'd see something like this at about the time you started hearing
audio:

0xYY - Probation passed - setting RTP source address to xxx.xxx.xxx.xxx

Asterisk drops RTP packets until it locks onto an RTP source. It does this
to prevent media injection attacks. The default probation period for an RTP
source is four packets - you can configure the probationary period as well
as whether or not strict RTP checking is enabled in rtp.conf.

Matt

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Voicemail greeting playback issues?

[Patrick Lists]

On 11/26/2013 12:24 AM, Doug Lytle wrote:
> Bryant Zimmerman wrote:
>> Hey all
>>
>> I believe I found the bug in Asterisk 11.xxx If someone can help me
>> verify it.
> 
> Actually,
> 
> I wouldn't consider it a bug.  I've know for years that you need to
> answer a channel before you play back audio or strange things can and
> will happen.

That's what I do since the 0.x days. IIRC in recent Asterisk versions
some apps answer before doing anything else. Guess the voicemail app is
not one of them. I always answer first followed by a small Wait and then
execute the app.

Regards,
Patrick

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Re: [asterisk-users] Voicemail greeting playback issues?

[Doug Lytle]

Bryant Zimmerman wrote:
> Hey all
>
> I believe I found the bug in Asterisk 11.xxx If someone can help me
> verify it.

Actually,

I wouldn't consider it a bug.  I've know for years that you need to
answer a channel before you play back audio or strange things can and
will happen.

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

From: "Bryant Zimmerman" 
Sent: Monday, November 25, 2013 2:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail greeting playback issues?


From: "Doug Lytle" 
Sent: Monday, November 25, 2013 2:01 PM
To: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial 
Discussion" 
Subject: Re: [asterisk-users] Voicemail greeting playback issues?

>> Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue 
goes away.

I don't see this under 11.5.1

Doug

---

Doug

Thank you for your response. It is good to hear that you are not having the 
issue. 
It gives me hope that there is a way to resolve this quickly. 

Do  you have an thing special around your voicemail configuration? We  
started with the 11.xx sample config and mapped our settings from  1.8.x.   
Both our 11.xx and 1.8.x systems are running on the same  virtual server. 
Both are reading and writing audio and vm files to and  from the local 
storage.  I forced off g729 to ensure that it was not  causing the issues.

Do you know of any way to force a higher level of debugging to see why the 
voicemail application would be having an issue?

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003



Hey all

I believe I found the bug in Asterisk 11.xxx If someone can help me verify 
it.

My voice mail test scripts do not answer or wait they just drop you into 
the voicemail box. 

It appears that something with Asterisk 11.xx is causing the voicemail() 
command to drop in and ether not play or mess up the prompts. If you have 
not given it at least one second in the channel before passing it to the 
voicemail() command.
If you throw a wiat(1) just before the voicemail() command the prompts play 
correctly. So if you have rung extensions using dial() before going to 
voicemail that appears to be enough time. 

If you place an inbound call directly to voicemail() with no pause then you 
have an issue. 

Example Broken:
exten => _9XXX,1,Set(l_VMExt=${EXTEN:1})
exten => _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext})
exten => _9XXX,n,GotoIf($["${VMBOXEXISTSSTATUS}"="FAILED"]?doHangup)
exten => _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u)
exten => _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup)
exten => _9XXX,n,Hangup()

Example Works:
exten => _9XXX,1,Set(l_VMExt=${EXTEN:1})
exten => _9XXX,n,MailboxExists(${l_VMExt}@${siteVMContext})
exten => _9XXX,n,GotoIf($["${VMBOXEXISTSSTATUS}"="FAILED"]?doHangup)
exten => _9XXX,n,Wait(1)
exten => _9XXX,n,Voicemail(${l_VMExt}@${siteVMContext},u)
exten => _9XXX,n(doHangup),NoOp(Issue 9XXX Hangup)
exten => _9XXX,n,Hangup()

The code that is broken with Asterisk 11.xx worked in Asterisk 1.8.x
Can anyone confirm this?

Thanks
Bryant Zimmerman()

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Re: [asterisk-users] Voicemail greeting playback issues?

[Bryant Zimmerman]

From: "Doug Lytle" 
Sent: Monday, November 25, 2013 2:01 PM
To: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial 
Discussion" 
Subject: Re: [asterisk-users] Voicemail greeting playback issues?

>> Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue 
goes away.

I don't see this under 11.5.1

Doug

---

Doug

Thank you for your response. It is good to hear that you are not having the 
issue. 
It gives me hope that there is a way to resolve this quickly. 

Do  you have an thing special around your voicemail configuration? We  
started with the 11.xx sample config and mapped our settings from  1.8.x.   
Both our 11.xx and 1.8.x systems are running on the same  virtual server. 
Both are reading and writing audio and vm files to and  from the local 
storage.  I forced off g729 to ensure that it was not  causing the issues.

Do you know of any way to force a higher level of debugging to see why the 
voicemail application would be having an issue?

Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


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Re: [asterisk-users] Voicemail greeting playback issues?

[Doug Lytle]

>> Both 11.2.1 and 11.6 do this. If I drop back to 1.8.current the issue goes 
>> away. 


I don't see this under 11.5.1 

Doug 


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Re: [asterisk-users] Voicemail Prepend Message Forwarding Not Working [SOLVED]

[Jr Richardson]

>
> Hi All,
>
> First I've heard of this feature not working from a customer.  I did some
> digging and this is a common bug in several older Asterisk versions, it has
> more than a few patches in the bug tracker.  I've tried a few of them but
> none will apply to a specific version I'm currently running for a customer,
> 1.6.0.28.
>
> Does anyone have a patch file that will apply to this version or an
> app_voicemail.c file that is already patched and will compile with this
> versions to fix this particular bug?
>
>
I patched app_voicemail.c manually from the patch file (revision 233691),
recompiled and now prepending voicemail works.

Thanks.

JR
-- 
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Re: [asterisk-users] Voicemail variables on email subject

[James Cloos]

> "RdSS" == Rafael dos Santos Saraiva  writes:

RdSS> emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR}

RdSS> Return:
RdSS> Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?=

That is a proper encoding for an SMTP mail header which is in utf8.

RdSS> Expected:
RdSS> Subject: 1504|12|"Teste - Rafael" <1570>|16

The sent header decodes to this string:

Subect: 1504|12|"Teste_-_Rafael"_<1570>|0:16

Note the colon from $VM_DUR (minutes:seconds).

MUAs are supposed to decode that.

-JimC
-- 
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Re: [asterisk-users] Voicemail variables on email subject

[A J Stiles]

On Monday 05 August 2013, Rafael dos Santos Saraiva wrote:
> Hi
> 
> I have a problem w/ voicemail, the subject message is corruption when used
> voicemail variables, e.g. :
> voicemail.conf
> emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR}
> 
> Return:
> Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?=
> 
> Expected:
> Subject: 1504|12|"Teste - Rafael" <1570>|16

That looks about sane for a subject line in UTF-8 encoding  (I haven't studied 
it too closely).  What mail client have you been using to retrieve the e-mail?  
Is it UTF-8 aware?

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Voicemail variables on email subject

[jg]

I checked your original message, and I guess the expected string was a little 
bit different:
1504|12|"Teste - Rafael" <1570>|0:16
I can't see anything wrong with quoted printable decoding. My best guess is still the email 
client and its settings.


jg

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Re: [asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

I noticed that the problem occurs when I use the variables "${VM_DUR}" and
"${VM_CALLERID}." Only the subject of the message, if the body is not the
problem. Using UTF or utf the same problem occurs.


Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956
*Digium Certified Asterisk Administrator (dCCA)*
http://www.astdocs.com | 


2013/8/6 jg 

> I checked the raw text of my voicemail messages today and I saw pretty
> much the same escape sequences for UTF-8 like you did, but I do not have
> any display problem. You could save the message locally and hand edit it
> (starting with uppercase UTF instead of lowercase utf).
>
> jg
>
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Re: [asterisk-users] Voicemail variables on email subject

[jg]

I checked the raw text of my voicemail messages today and I saw pretty much the same escape 
sequences for UTF-8 like you did, but I do not have any display problem. You could save the 
message locally and hand edit it (starting with uppercase UTF instead of lowercase utf).


jg

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Re: [asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

When sending by SendMail the problem is the same in any email client.


Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956
*Digium Certified Asterisk Administrator (dCCA)*
http://www.astdocs.com | 


2013/7/25 jg 

> If I read your mailcmd correctly you are not really mailing but just
> dumping the data. Is the display correct when you use the default setting
> "/usr/sbin/sendmail -t"? You could send the mail to a local account and
> open it with mutt.
>
> jg
>
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Re: [asterisk-users] Voicemail variables on email subject

[jg]

Some time ago I had a similar problem but it turned out to be a display problem of the email 
client (Outlook Express on an old XP machine).


jg

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Re: [asterisk-users] Voicemail variables on email subject

[jg]

If I read your mailcmd correctly you are not really mailing but just dumping the data. Is the 
display correct when you use the default setting "/usr/sbin/sendmail -t"? You could send the 
mail to a local account and open it with mutt.


jg

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Re: [asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

I tried with utf-8, iso8859-1 and us-ascii.

I used the Sendmail client, but now testing with "mailcmd=cat >
/tmp/voicemail.txt"

The version of Asterisk is 1.8.22.0.


Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956
*Digium Certified Asterisk Administrator (dCCA)*
http://www.astdocs.com | 


2013/7/25 jg 

> What is the value of "charset" in voicemail.conf?
>
> Have you tried a different Email client?
>
> jg
>
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Re: [asterisk-users] Voicemail variables on email subject

[jg]

What is the value of "charset" in voicemail.conf?

Have you tried a different Email client?

jg

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Re: [asterisk-users] voicemail customization

[Steve Edwards]

On Tue, 5 Feb 2013, Darin Iv wrote:

Is there any solution, I want to know at first sight from which division 
the voicemail came from like for example I need 3 more users to send 
when some one calls an inbound route named darin when it reaches darin 
voicemail then admin should send from da...@yahoo.com


I'm not sure what you want...

I'm guessing you want to send voicemail as email and set the 'from' 
address based on the inbound route.


In voicemail.conf, you can set the program used to send email. This could 
be a 'wrapper' for looking up the 'from' address and constructing a 
command line to execute either sendmail or sendEmail.pl or whatever.


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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Voicemail to text for Asterisk

[Carlos Alvarez]

On Mon, Oct 22, 2012 at 2:43 PM, Nickolay V. Shmyrev
wrote:

> There is no holy grail yet, speech technology deployment requires a
> close cooperation between the speech technology provider and the users.
> It's not plug and play but after some joint efforts automated
> transcriptions must be useful.
>
> If anyone wants to experiment with CMUSphinx-based automated solution to
> transcribe voicemails, drop me a note. The results could be pretty
> interesting.
>
>
We will probably give it a try, though not sure when.  Probably in about a
month.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Voicemail to text for Asterisk

[Nickolay V. Shmyrev]

On 22/10/2012 at 16:02 -0400, Bryant Zimmerman wrote:
> Carlos
> 
> I have tried several solutions and non of them have been worth the
> money. I have worked with transcription companies and they are the
> best but they are expensive. If you do find something that works let
> the groups know as there are a few of us out here that are looking for
> that holy grail of speech to text. 

There is no holy grail yet, speech technology deployment requires a
close cooperation between the speech technology provider and the users.
It's not plug and play but after some joint efforts automated
transcriptions must be useful.

If anyone wants to experiment with CMUSphinx-based automated solution to
transcribe voicemails, drop me a note. The results could be pretty
interesting.



signature.asc
Description: This is a digitally signed message part
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Re: [asterisk-users] Voicemail to text for Asterisk

[Christopher Harrington]

On Mon, Oct 22, 2012 at 3:05 PM, Lefteris Zafiris  wrote:

> If you are able to find a reliable way of chopping speech samples in
> segments no bigger
> than 20 seconds based on silence detection, so words wont be cut in half,
> you might come
> up with something very similar to Google Voice transcription service.
>

Unfortunately Google's transcription is vastly improved by its context
comprehension (for instance, understanding that the word "phone" is likely
to be followed by words like "call" or "number") and chopping up the audio,
even between words, will reduce that context data for the transcriber.

Good luck, anyway.

-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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Re: [asterisk-users] Voicemail to text for Asterisk

[Lefteris Zafiris]

On Mon, 22 Oct 2012 12:47:51 -0700
Carlos Alvarez  wrote:
> 
> In-house transcriptions are definitely out of the question, but any
> experience with outsourced solutions would be useful.  As far as I can tell
> the current service is automated, and as awful as Google Voice, yet they
> find it useful.  Their existing carrier uses Broadsoft and I'm not sure if
> they have that built in.
> 

Voice recognition for asterisk based on Google speech API is already 
available[1],
the problem with this service is that it's limited to 20-30 seconds of speech 
data,
which isn't suitable for transcripting voicemails.
If you are able to find a reliable way of chopping speech samples in segments 
no bigger
than 20 seconds based on silence detection, so words wont be cut in half, you 
might come
up with something very similar to Google Voice transcription service.
But I would recommend against using this into production since google haven't 
yet
defined the terms of service for speech recognition, and its more or less a 
hack for
now.

[1] http://zaf.github.com/asterisk-speech-recog/


Lefteris Zafiris

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Re: [asterisk-users] Voicemail to text for Asterisk

[Bryant Zimmerman]

Carlos

I have tried several solutions and non of them have been worth the money. I 
have worked with transcription companies and they are the best but they are 
expensive. If you do find something that works let the groups know as there 
are a few of us out here that are looking for that holy grail of speech to 
text. 

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 


 From: "Carlos Alvarez" 
Sent: Monday, October 22, 2012 3:49 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Voicemail to text for Asterisk

On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington  
wrote:
 All automated solutions -- paid or free -- are terrible. The technology 
simply does not exist at this point at a level that is acceptable to most 
customers. If quality is paramount, you are better off doing the 
transcription in-house with a human.  
 In-house transcriptions are definitely out of the question, but any 
experience with outsourced solutions would be useful.  As far as I can tell 
the current service is automated, and as awful as Google Voice, yet they 
find it useful.  Their existing carrier uses Broadsoft and I'm not sure if 
they have that built in. 
  -- 
Carlos Alvarez TelEvolve 602-889-3003 


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Re: [asterisk-users] Voicemail to text for Asterisk

[Carlos Alvarez]

On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington wrote:

> All automated solutions -- paid or free -- are terrible. The technology
> simply does not exist at this point at a level that is acceptable to most
> customers. If quality is paramount, you are better off doing the
> transcription in-house with a human.
>

In-house transcriptions are definitely out of the question, but any
experience with outsourced solutions would be useful.  As far as I can tell
the current service is automated, and as awful as Google Voice, yet they
find it useful.  Their existing carrier uses Broadsoft and I'm not sure if
they have that built in.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Voicemail to text for Asterisk

[Christopher Harrington]

On Mon, Oct 22, 2012 at 2:16 PM, Carlos Alvarez wrote:

> A customer has asked us to provide that feature.  I know there are a few
> methods and products out there, but I haven't paid attention in a while.
>  It is for about 300 users, and we'll consider open as well as paid-for
> products.  We would prefer to pay for supported products as the cost will
> be passed on to the customer and they are willing to pay for quality.  Do
> not want any complex scripting screwing around with third parties and such.
>  Your ideas welcome.
>
> All automated solutions -- paid or free -- are terrible. The technology
simply does not exist at this point at a level that is acceptable to most
customers. If quality is paramount, you are better off doing the
transcription in-house with a human.


-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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Re: [asterisk-users] Voicemail to text for Asterisk

[Danny Nicholas]

Unless I missed something, there isn't anything out there that is as cheap
or reliable as human translation in this case.  If I did miss it, I know
somebody will correct me.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Monday, October 22, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail to text for Asterisk

 

A customer has asked us to provide that feature.  I know there are a few
methods and products out there, but I haven't paid attention in a while.  It
is for about 300 users, and we'll consider open as well as paid-for
products.  We would prefer to pay for supported products as the cost will be
passed on to the customer and they are willing to pay for quality.  Do not
want any complex scripting screwing around with third parties and such.
Your ideas welcome.

 

-- 

Carlos Alvarez

TelEvolve

602-889-3003

 

 

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Re: [asterisk-users] Voicemail not working with vm boxes named with a star

[Karsten Wemheuer]

Hi Matthew,

Am Donnerstag, den 20.09.2012, 06:27 -0500 schrieb Matthew Jordan:
> - Original Message -
> > From: "Karsten Wemheuer" 
> > To: asterisk-users@lists.digium.com
> > Sent: Thursday, September 20, 2012 2:28:07 AM
> > Subject: [asterisk-users] Voicemail not working with vm boxes named with a  
> >   star
> > 
> > Hi list,
> > 
> > in asterisk 1.4 and maybe earlier it was possible to use voicemail
> > system with mailboxes starting with some special characters like *.
> > The
> > line in voicemail.conf was like this:
> > *123 => , AB,,,tz=cet|attach=no|
> > 
> > Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4.
> > 
> > In Asterisk 1.8 the above scenario is not working any more. The
> > Voicemail application reports an error message:
> > WARNING: app_voicemail.c: leave_voicemail: No entry in voicemail
> > config
> > file for '*123'
> > 
> > Is this a known bug, fixed in newer versions (I currently use 1.8.11)
> > or
> > should I file a bug report?
> 
> Nope, this is not a bug.  The change in behavior was deliberate (see
> https://issues.asterisk.org/jira/browse/ASTERISK-17433).  Starting a mailbox
> with a '*' conflicted with the auto-attendant feature in app_voicemail, 
> wherein
> a user can be redirected to the 'a' extension by sending the '*' DTMF.  There
> were a number of weird side effects that occurred due to this, most of which
> involved users who had created a mailbox that began with a '*' unable to check
> their voicemail.
> 
> Since it would be very easy to mis-configure your system if you had both the
> auto-attendant feature enabled and allowed users to have a mailbox/password 
> that
> started with '*', we prevented the latter scenario.

Thank You for Your response. I got Your message after writing my second
message. 

I understand the problem. But if the issue is triggered mixing boxes
starting with * and the assistant feature, what about the operator
feature? Should boxes starting with '0' also be prevented?

Thanks,

Karsten



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Re: [asterisk-users] Voicemail not working with vm boxes named with a star

[Karsten Wemheuer]

Hi list,

Am Donnerstag, den 20.09.2012, 09:28 +0200 schrieb Karsten Wemheuer:
> Hi list,
> 
> in asterisk 1.4 and maybe earlier it was possible to use voicemail
> system with mailboxes starting with some special characters like *. The
> line in voicemail.conf was like this:
> *123 => , AB,,,tz=cet|attach=no|
> 
> Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4.
> 
> In Asterisk 1.8 the above scenario is not working any more. The
> Voicemail application reports an error message:
> WARNING: app_voicemail.c: leave_voicemail: No entry in voicemail config
> file for '*123'
> 
> Is this a known bug, fixed in newer versions (I currently use 1.8.11) or
> should I file a bug report?

After looking at log files and source code, I found out, that in
function "find_or_create" in app_voicemail.c there is a statement:

if (!ast_strlen_zero(box) && box[0] == '*') {

right at the beginning of that function. This leads to not setting up a
mailbox *123, whereas "123*" is allowed. The logging says:
The '*' character, when it is the first character in a mailbox
or password, is used to jump to a predefined extension 'a'. A
mailbox or password beginning with '*' is not valid and will be
ignored.

I do not see, why a mailbox should not be valid starting with '*'. The
feature to jump to a predefined extension by pressing * exists in
Asterisk 1.4 but you can create a mailbox starting with '*' in 1.4. If
for some reason the feature to jump to an predefined extension by
pressing some key forbids using that key as first part of a mailbox,
than the above code should prevent using '0' as first part too, I think.
Pressing '0' is analog to pressing '*' according to the documentation.

Does anybody know, why the "if"-statement is put in the code?

Thanks,

Karsten



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Re: [asterisk-users] Voicemail not working with vm boxes named with a star

[Matthew Jordan]

- Original Message -
> From: "Karsten Wemheuer" 
> To: asterisk-users@lists.digium.com
> Sent: Thursday, September 20, 2012 2:28:07 AM
> Subject: [asterisk-users] Voicemail not working with vm boxes named with 
> astar
> 
> Hi list,
> 
> in asterisk 1.4 and maybe earlier it was possible to use voicemail
> system with mailboxes starting with some special characters like *.
> The
> line in voicemail.conf was like this:
>         *123 => , AB,,,tz=cet|attach=no|
> 
> Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4.
> 
> In Asterisk 1.8 the above scenario is not working any more. The
> Voicemail application reports an error message:
> WARNING: app_voicemail.c: leave_voicemail: No entry in voicemail
> config
> file for '*123'
> 
> Is this a known bug, fixed in newer versions (I currently use 1.8.11)
> or
> should I file a bug report?

Nope, this is not a bug.  The change in behavior was deliberate (see
https://issues.asterisk.org/jira/browse/ASTERISK-17433).  Starting a mailbox
with a '*' conflicted with the auto-attendant feature in app_voicemail, wherein
a user can be redirected to the 'a' extension by sending the '*' DTMF.  There
were a number of weird side effects that occurred due to this, most of which
involved users who had created a mailbox that began with a '*' unable to check
their voicemail.

Since it would be very easy to mis-configure your system if you had both the
auto-attendant feature enabled and allowed users to have a mailbox/password that
started with '*', we prevented the latter scenario.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Voicemail full.

[Luis H. Forchesatto]

Yep, maxmsg is set to some value and it reached. To make things work again
i've moved que messages to a new directory and voicemail is working now.

2012/8/3 Steve Edwards 

> Un-top-posting...
>
>
> On Fri, 3 Aug 2012, Luis H. Forchesatto wrote:
>
>  I've made a call to our elastix server and the call was redirected to the
>> voicemail, which the user should leave a message. Intead recording the call
>> the service returned a message like "Sorry, but the user's mailbox can't
>> accept more messages". I'm a little lost in the configs here, what
>> parameter should I edit to increase the mailbox capacity or what steps I
>> take to 'clean' the mailbox?
>>
>
> On Fri, 3 Aug 2012, Danny Nicholas wrote:
>
>  Looking at my voicemail.conf I note this snippet:
>>
>> ; Maximum number of messages per folder. If not specified, a default value
>>
>> ; (100) is used. Maximum value for this option is .
>>
>> ;maxmsg=100
>>
>> So in my case max messages is .
>>
>
> (After a quick glance at the source...)
>
> If not specified, the limit would be MAXMSG (100) not MAXMSGLIMIT ().
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>



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Re: [asterisk-users] Voicemail full.

[Steve Edwards]

Un-top-posting...

On Fri, 3 Aug 2012, Luis H. Forchesatto wrote:

I've made a call to our elastix server and the call was redirected to 
the voicemail, which the user should leave a message. Intead recording 
the call the service returned a message like "Sorry, but the user's 
mailbox can't accept more messages". I'm a little lost in the configs 
here, what parameter should I edit to increase the mailbox capacity or 
what steps I take to 'clean' the mailbox?


On Fri, 3 Aug 2012, Danny Nicholas wrote:


Looking at my voicemail.conf I note this snippet:

; Maximum number of messages per folder. If not specified, a default 
value


; (100) is used. Maximum value for this option is .

;maxmsg=100

So in my case max messages is .


(After a quick glance at the source...)

If not specified, the limit would be MAXMSG (100) not MAXMSGLIMIT ().

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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Voicemail full.

[Danny Nicholas]

Looking at my voicemail.conf I note this snippet:

; Maximum number of messages per folder.  If not specified, a default value

; (100) is used.  Maximum value for this option is .

;maxmsg=100

So in my case max messages is .  

 

Assuming you are storing your message in files instead of a Database, you
would look at /var/spool/asterisk/voicemail/default/xxx/INBOX where xxx is
replaced by the mailbox number.  As I understand it, you would have to
renumber the files from msg to the maximum number and there are 2 to 4
files per message depending on your setup.  I think there was a thread on
this in  July 2012.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis H.
Forchesatto
Sent: Friday, August 03, 2012 7:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail full.

 

Hi

 

I've made a call to our elastix server and the call was redirected to the
voicemail, which the user should leave a message. Intead recording the call
the service returned a message like "Sorry, but the user's mailbox can't
accept more messages". I'm a little lost in the configs here, what parameter
should I edit to increase the mailbox capacity or what steps I take to
'clean' the mailbox?


 

-- 
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Re: [asterisk-users] Voicemail Emails

[Kevin P. Fleming]

On 07/20/2012 04:48 PM, Warren Selby wrote:

On Fri, Jul 20, 2012 at 12:53 PM, Josh Hopkins mailto:j...@prorivertech.com>> wrote:

Has anyone been able to make an html template for the voicemail
emails. We would love to customize them beyond just plain text. We
have dome some Google searches and have not been able to come up
with much. 

__ __

I believe that Switchvox has customized the voicemail email  into
html.  Has anyone ever tried this?  Thanks,

 /Josh



What about changing 'mailcmd=' to a shell script that rewrites the email
in the format you want before sending it to sendmail?


That is most likely the best way to accomplish it; it would need to 
receive the already-composed email on stdin, then parse it, modify it, 
and regenerate it before sending it to the real mailer. This could be 
done using standard email libraries in many scripting languages.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Voicemail Emails

[Warren Selby]

On Fri, Jul 20, 2012 at 12:53 PM, Josh Hopkins wrote:

> Has anyone been able to make an html template for the voicemail emails. We
> would love to customize them beyond just plain text. We have dome some
> Google searches and have not been able to come up with much. 
>
> ** **
>
> I believe that Switchvox has customized the voicemail email  into html.
> Has anyone ever tried this?  Thanks,
>
> /Josh
>
>
>
What about changing 'mailcmd=' to a shell script that rewrites the email in
the format you want before sending it to sendmail?

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] Voicemail Emails

[Danny Nicholas]

You can change voicemail.conf to use a delivery option other than sendmail.
Other than that, show us your body line from voicemail.conf

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Hopkins
Sent: Friday, July 20, 2012 4:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Emails

 

Yes and you just get html code in the email rather than the html format.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi
Sent: Friday, July 20, 2012 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Emails

 

Have you tried to insert the HTML code directly into the body?


Il 20/07/12 19:53, Josh Hopkins ha scritto:

Has anyone been able to make an html template for the voicemail emails. We
would love to customize them beyond just plain text. We have dome some
Google searches and have not been able to come up with much. 

 

I believe that Switchvox has customized the voicemail email  into html.  Has
anyone ever tried this?  Thanks,

/Josh

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Re: [asterisk-users] Voicemail Emails

[Kevin P. Fleming]

On 07/20/2012 04:35 PM, Josh Hopkins wrote:

Yes and you just get html code in the email rather than the html format.


app_voicemail does not send MIME-encapsulated emails, it sends raw 
email. Unless you could add a Content-Type header to the message (which 
app_voicemail doesn't allow you to do), the email client that receives 
the HTML is going to treat it as plain text, which is what you saw.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Voicemail Emails

[Josh Hopkins]

Yes and you just get html code in the email rather than the html format.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi
Sent: Friday, July 20, 2012 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Emails

Have you tried to insert the HTML code directly into the body?

Il 20/07/12 19:53, Josh Hopkins ha scritto:
Has anyone been able to make an html template for the voicemail emails. We 
would love to customize them beyond just plain text. We have dome some Google 
searches and have not been able to come up with much.

I believe that Switchvox has customized the voicemail email  into html.  Has 
anyone ever tried this?  Thanks,
/Josh
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Re: [asterisk-users] Voicemail Emails

[Danilo Dionisi]

Have you tried to insert the HTML code directly into the body?

Il 20/07/12 19:53, Josh Hopkins ha scritto:


Has anyone been able to make an html template for the voicemail 
emails. We would love to customize them beyond just plain text. We 
have dome some Google searches and have not been able to come up with 
much.


I believe that Switchvox has customized the voicemail email  into 
html.  Has anyone ever tried this? Thanks,


/Josh

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Re: [asterisk-users] Voicemail attachment format

[Warren Selby]

On Mon, Jun 25, 2012 at 9:23 AM, khalid touati wrote:

> Hi All,
> I have a simple urgent question that I couldn't find the answer yet, can
> we customize the voicemail attachment format *per user* in asterisk *1.2 
> *(like
> all receive wav attch but one or two users receive attch in gsm format)? if
> yes can you show me how please?
>
>

I don't think that was an option in 1.2, but I haven't used 1.2 in so long
I may be off.  Hopefully one of our resident 1.2 luddite's will see this
and have a more definitive answer for you.

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http://www.SelbyTech.com 
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Re: [asterisk-users] voicemail password with phone instrument

[Kamlesh Kumar]

still waiting for valuable reply on this. Regards,Kamlesh
 From: kamlesh_...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 15 Jun 2012 12:19:48 +
Subject: [asterisk-users] voicemail password with phone instrument








Hello,
 
voicemail password is not getting changed through phone handset while IVR 
indicates that password has been changed. During google I found that uniqueid 
column must not be changed so it is not changed. Please guide on this.
 
During debug log I found below but in mysql db new password is not getting 
updated,
 
[Jun 15 13:54:07] VERBOSE[6418] file.c: --  Playing 
'vm-newpassword.gsm' (language 'en')
[Jun 15 13:54:10] DEBUG[6418] channel.c: Set channel SIP/123-0005 to write 
format ulaw
[Jun 15 13:54:15] VERBOSE[6418] file.c: --  Playing 
'vm-reenterpassword.gsm' (language 'en')
[Jun 15 13:54:22] DEBUG[6418] app_voicemail.c: User 123 set password to  of 
length 4
[Jun 15 13:54:22] DEBUG[6418] channel.c: Set channel SIP/123-0005 to write 
format gsm
[Jun 15 13:54:22] VERBOSE[6418] file.c: --  Playing 
'vm-passchanged.gsm' (language 'en')

Regards,
Kamlesh
 
 
  

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