On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl wrote:
> Hi all,
>
> I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the
> 1.8.x to 10.4.x upgrade was painful; some of the modules had been
> renamed, if I recall correctly.
>
> So, is there a list of MAJOR changes and GOTCHA's between 10
On 07/10/2013 01:04 PM, bilal ghayyad wrote:
Hello;
To let the Phone answer automatically, this can be configured from
asterisk (at the sip.conf for the phone)? Or it has to be from the IP
Phone? Because, some phones does not support auto answer, also we do not
need to do it for each Phone.
De
OK, thanks for the advice. No, there's no filter so I'll look into that.
On Wed, Jul 10, 2013 at 3:02 PM, Patrick Lists
wrote:
> On 07/10/2013 06:46 PM, Chris Gentle wrote:
> [snip]
>
>> and then others can connect via SIP. For some reason, when the
>> speaker says words with S's and F's, they
Hi all,
I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the
1.8.x to 10.4.x upgrade was painful; some of the modules had been
renamed, if I recall correctly.
So, is there a list of MAJOR changes and GOTCHA's between 10.x and
11.x? I'm hoping for something a little less granular tha
Okay, so I is no good. Does anybody else have a work-around for this?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 10, 2013 1:43 PM
To: Asterisk Users Mailing Lis
"I" has the same limitations as dialplan timeouts, you have to be in a
Background or WaitExten or similar for them to work.These items are
designed for IVRS.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf O
It seems likely that this is exactly what is happening. I'd rather not change
the code though, but rather fix the dialplan. I'm thinking using the 'i'
extension would work just the same - would there be a reason to use a wildcard
pattern match instead of i?
-Justin
-Original Message-
On Wed, Jul 10, 2013 at 3:11 PM, Eric Wieling wrote:
> From chan_dahdi.c, don't know if it applies to your situation or not.
>
> /*! \brief Wait up to 16 seconds for first digit (FXO logic) */
> static int firstdigittimeout = 16000;
>
> /*! \brief How long to wait for following digits (FXO logic)
>From chan_dahdi.c, don't know if it applies to your situation or not.
/*! \brief Wait up to 16 seconds for first digit (FXO logic) */
static int firstdigittimeout = 16000;
/*! \brief How long to wait for following digits (FXO logic) */
static int gendigittimeout = 8000;
/*! \brief How long to w
On 07/10/2013 06:46 PM, Chris Gentle wrote:
[snip]
and then others can connect via SIP. For some reason, when the
speaker says words with S's and F's, they almost sound distorted. Not
quite static but you can tell the quality has been affected. May just
be a side-effect of 8,000 Hz. Just wond
So then, by saying "If the digits already dialed match an extension in the
dialplan...wait 3 seconds...", then we're saying that asterisk has found a
match, and the match is the bad-extension. Here is the bad-number context that
is included:
[bad-number]
include => bad-number-custom
exten => _
Hello Andy,
Have you tried using SetMusicOnHold command before Queue command?
BR,
Ioan
On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas wrote:
> Hi All,
>
> Sorry if this has been covered already, but I don't tend to follow this
> list as close as I should these days.
>
> Problem is that if a c
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen <
jkil...@allamericanasphalt.com> wrote:
> I have an installation that has analog phones connected via T1 channel
> banks. I’m getting complaints from users that they will enter a partial
> number (eg 91213), then turn away to get the next few digit
Hello;
To let the Phone answer automatically, this can be configured from asterisk (at
the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some
phones does not support auto answer, also we do not need to do it for each
Phone.
Regards
Bilal--
___
Hi All,
Sorry if this has been covered already, but I don't tend to follow this
list as close as I should these days.
Problem is that if a call comes in to a queue without option 'r'
specified - moh plays as expected. Now, when that call is answered, all
is fine. Trouble comes when that person t
I believe the TIMEOUT() function and apps only work once you are in an IVR or
other dialplan application which waits for digits.On DAHDI channels I think
you have to modify the source code if you want to change the timeout when
dialing from a dialtone.
-Original Message-
From: aster
Okay, after enabling DTMF logging, what I see is a handset being picked up, 7
digits being pressed in 4 seconds, and then 3 seconds input is determined to be
done and the call is processed (to the catch-all 'bad-number').
What I don't understand is that if the digit timeout is set to 5, then why
On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth wrote:
> The sampling frequency for u-law is 8,000 Hz. You can't produce a recording
> with higher quality than the source, so you'd have to switch to a wideband
> codec
> to improve the conferences and recordings [1] [2].
OK, thanks for the info
Values for the timeouts just before the 'cannot complete as dialed, please try
your call again':
absolute: 0
digit: 5.000
response: 10.000
I've enabled DTMF logging to try to get a better log for interpretation.
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi Eduardo
Le 10/07/2013 14:30, Eduardo Leones a écrit :
I have a question about global variables. Is it possible to somehow
keep global variables unset via Dial Plan even Restarting asterisk?
[...]
From extensions.conf version 1.8
; If clearglobalvars is set, global variables will be clear
Chris Gentle wrote:
>>>
>>> Is there any way I can improve the audio quality in a confbridge in
>>> Asterisk 11? I've changed the internal_sample_rate setting to 44100
>>> but that doesn't seem to make any difference. I would also think this
>>> would make my confbridge recordings 44100 but they
ulaw
On Wed, Jul 10, 2013 at 7:40 AM, basteon wrote:
> Hi,
> What codec do you use with yours subscribers?
>
>
> On 9 July 2013 23:45, Chris Gentle wrote:
>>
>> Is there any way I can improve the audio quality in a confbridge in
>> Asterisk 11? I've changed the internal_sample_rate setting to 4
Hi,
What codec do you use with yours subscribers?
On 9 July 2013 23:45, Chris Gentle wrote:
> Is there any way I can improve the audio quality in a confbridge in
> Asterisk 11? I've changed the internal_sample_rate setting to 44100
> but that doesn't seem to make any difference. I would also
I have a question about global variables. Is it possible to somehow keep
global variables unset via Dial Plan even Restarting asterisk?
tks
Eduardo
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-Original Message-
From: asterisk-users-requ...@lists.digium.com
Sent: 7/10/2013 1:00 AM
To: asterisk-users@lists.digium.com
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