[asterisk-users] IVR extension dialing error
Danny Nicholas
danny at debsinc.com
Fri Jun 18 08:16:10 CDT 2010
I would definitely change the prompt from 1 to 0. It is not an advisable
practice to have an IVR selection that can be misinterpreted like this.
Assuming that all of your extensions are in 1000-1999, 2 for the operator
would be just as good; the important thing is that you don't have a single
digit extension 1.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Friday, June 18, 2010 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IVR extension dialing error
Hi, I tell you I've made some calls from a land-phone to my IVR in
order to avoid the possible poor quality of cell phone's DTMF, and
when I called extension 1003 I was connected to extension 1000
again....the same error.
My IVR says "dial 1 to connect to operator or dial the extension in
case you know".....and my extension ranges is 1000-1999, so I think it
could be a problem that extensions and IVR option start with the same
digit: 1.
When I'll be at work I'm thinking in modify the IVR speech in order to
say "dial 0 to connect to operator.....", and not "dial 1 to connect
to operator....", so IVR option and extensions will not start with the
same digit.
Do you think this may be the problem ???
Thanks a lot and sorry for my interruption.
Alejandro
2010/6/17 Danny Nicholas <danny at debsinc.com>:
> According to this link
> http://www.smallnetbuilder.com/content/view/30469/82/1/2/
>
> You probably want to make 80 be 120. This is a millisecond delay value, so
> the 500 value is a "give it up" proposition; 200 might be doable for your
> outliers.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alejandro
> Cabrera Obed
> Sent: Thursday, June 17, 2010 12:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IVR extension dialing error
>
> OK, now I understand..but just one more question...In the DTMF
> settings tab from the GSM gateway manager I have this line:
>
> Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms
>
> What does this setting really mean and do I have to modify the current
value
> ???
>
> Final thanks :)
>
> 2010/6/17 Zeeshan Zakaria <zishanov at gmail.com>:
>> I once setup a callback system for someone and we had these DTMF issues
on
>> constant basis, and all the complains were from cell phone users. At that
>> time I found out that even my own cellphone would not DTMF correctly from
>> certain locations, including my home, but would work perfectly fine from
> my
>> work location. Probably times of the day matters too, but yes, calling
> from
>> cell phones does result in DTMF issues, and the reason is that it is just
>> the audio signals, which get distorted based on various factors like the
>> signal strength, cell tower transmission quality, transcodings, etc.
>>
>> Zeeshan A Zakaria
>>
>> --
>> www.ilovetovoip.com
>>
>> On 2010-06-17 11:25 AM, "Alejandro Cabrera Obed" <aco1967 at gmail.com>
> wrote:
>>
>> Danny, so you say it's a problem of the cell phone and not the
>> Astreisk or GSM Gateway ???
>>
>> OK, in this case if I call from a fixed phone (not a cell phone) to
>> the IVR, the DTMF quality problem will not be present....this may be a
>> good test, isn't it ??? Or do you suggest another test I can implement
>> ???
>>
>> Thanks again
>>
>> Alejandro
>>
>> 2010/6/17 Danny Nicholas <danny at debsinc.com>:
>>
>>> The physical location of the phone (access to towers) can vastly affect
>>> the
>>> quality of DTMF pass...
>>
>> --
>> Alejandro Cabrera Obed
>> aco1967 at gmail.com
>> www.alejandrocabrera.com.ar
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocati...
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>
>
>
> --
> Alejandro Cabrera Obed
> aco1967 at gmail.com
> www.alejandrocabrera.com.ar
>
> --
> _____________________________________________________________________
> -- 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
>
>
> --
> _____________________________________________________________________
> -- 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
>
--
Alejandro Cabrera Obed
aco1967 at gmail.com
www.alejandrocabrera.com.ar
--
_____________________________________________________________________
-- 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
More information about the asterisk-users
mailing list