Found the problem! In voicemail.conf, there is a setting called ‘exitcontext’
It was set to ‘vm-operator’. I found this on the voip-info.org site:
exitcontext
Optional context to drop the user into after he/she has pressed * or 0 to exit
voicemail. If not set, pressing * or 0 will return th
it definitely works while playing the greeting...
-- Executing [s@macro-ringphone:207] CELGenUserEvent("SIP/4999-0001",
"VMSCOVER,1503518585.1,4001") in new stack
-- Executing [s@macro-ringphone:208] VoiceMail("SIP/4999-0001",
"4001@default,u") in new stack
> 0x9af5428
Alix 6f2 = 2 LAN / 1 miniPCI / 1
miniPCI Express / LX800 / 256 MB / USB / dual SIM socket this is what I'm
using for my pbx.
Thanks all
Sent from my iPhone
Begin forwarded message:
> From: Roberto Rivera
> Date: August 23, 2017 at 4:43:31 PM EDT
> To: AstLinux Users Mailing List
> Subje
> Hi all,
> I was using astlinux then got a msg that I had no memory so it stopped
> everything. I rebooted and everything was fineit happened again but now I
> can't get access to the site for the past day it soI can make outbound
> calls and was getting ready to work on getting inbound
Tim,
For testing you might try also adding the 'd' option to VoiceMail()
--
d - Accept digits for a new extension in context c, if played during the
greeting. Context defaults to the current context.
--
try "1" first then "*" .
https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail
>F
I pressed ‘*’ twice while listening to my unavailable greeting, nothing
happened.
I believe Asterisk is doing nothing with the ‘*’:
-- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-0046",
"9373506524,u") in new stack
[Aug 23 15:50:32] DEBUG[1723][C-0052]: app_voic
That tells you that Asterisk is detecting the tone. Doesn't tell you what
it is doing with it... so you still need to trace dialplan execution (turn
off debug, leave verbose on) to see what action it is taking on the tone.
David
On Wed, Aug 23, 2017 at 12:13 PM, Tim Turpin wrote:
> *I won’t co
I won’t copy in the entire session (way too much info), but here’s the result
of my pressing *,*,1,2,3,4,5,6,#. It looks as though Asterisk is seeing the
DTMF.
[Aug 23 11:58:47] DEBUG[1557][C-0048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72
Check that the * key is not being captured for some other purpose (grep
into other .conf files). Check that you can match the * key outside of
voicemail... use WaitExten() and validate that your dialplan sees that.
You can also go into the asterisk console ("asterisk -r") and turn on
verbose and d
If I change my config to direct the call to VoiceMailMain(), I can log in
with DTMF digits, so I know the carrier is passing tones. And Asterisk is
recognizing them.
Thanks.
-Original Message-
From: Lonnie Abelbeck [mailto:li...@lonnie.abelbeck.com]
Sent: Wednesday, August 23, 2017 10:5
Tim,
Make sure in your sip.conf for your inbound provider the setting for "dtmfmode"
matches what your provider requires, Asterisk defaults to rfc2833 .
Lonnie
On Aug 23, 2017, at 9:20 AM, Tim Turpin wrote:
> Getting closer, I think.
>
> I’m starting to wonder if the DTMF ‘*’ is being reco
Getting closer, I think.
I’m starting to wonder if the DTMF ‘*’ is being recognized at all. Now the
caller is dropped into the proper mailbox, but pressing ‘*’ does nothing.
Here’s extensions.conf:
[inbound]
exten => _NXXNXX,1,Answer
exten => _NXXNXX,n,NoOp(inbound-phone-cal
Also, be aware that voip-info has a lot of pretty outdated information as
it has not been updated for more recent versions of Asterisk. So while
google is great at finding info and you often land at voip-info, I
recommend also cross referencing to the master Asterisk Wiki at
https://wiki.asterisk.
set a variable first... the issue is that ${EXTEN} changes to 'a' when you *
out... ${EXTEN} is the current extension you are workign with and you want to
go to the original dialed extension.
[inbound]
exten => _NXXNXX,1,Answer
exten => _NXXNXX,n,NoOp(inbound-phone-call)
; set a variabl
This appears to possibly work for one mailbox user. We have a couple thousand
users, all dialing in via DID, and the process needs to be the same for all
users. My current extensions.conf looks like this:
[inbound]
exten => _NXXNXX,1,Answer
exten => _NXXNXX,n,NoOp(inbound-phone-call
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