-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: 08 July 2015 10:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
Indeed, thanks.
I'll let you know how it goes
'
Subject: Re: [asterisk-users] DTMF issue
You probably have to reload asrerisk after making the change.
Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
Jamie Rees jr...@gmlnt.com 7/7/2015 3:53 PM
Ah I see, in theory it's
,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue
It's called DTMF Talk-off. We have it too
It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile
phones but it happens at random on many external calls. If this happens to you,
especially on voice peaks (when the outside party said a particularly loud
syllable) then you probably have DTMF talk-off.
I think it's
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue
It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls
Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport.
See this discussion, which I found after I posted my first response:
http
: [asterisk-users] DTMF issue
In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport.
See this discussion, which I found after I posted my first response:
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Particularly this sentence
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: Monday, July 06, 2015 5:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue
Hello folks,
We have an issue with several Cisco SPA512G phones
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance.
I've tried changing the DTMF
On 7/6/15 5:53 PM, Jamie Rees wrote:
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an
Asterisk platform where several users hear loud, random beeps during
calls to external recipients. The noises are akin to button press
tones, are very loud and a significant
Up?
2012/8/20 Luis H. Forchesatto luisforchesa...@gmail.com
Thanks for your answer.
The logs where posted at pastebin, here the links:
- Working Phone: http://pastebin.com/q3pHcwna
- Not working phone: http://pastebin.com/iiCHPMmn
2012/8/20 Rusty Newton rnew...@digium.com
On 8/20/2012
Hi
I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA
on the network who autenticate the phones: Linksys PAP2,
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP
server at the same network all with g729 codecs and rfc2833 for the DTMF.
Making calls via
On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
Hi
I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of
ATA on the network who autenticate the phones: Linksys PAP2,
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the
VoIP server at the same network all with
Thanks for your answer.
The logs where posted at pastebin, here the links:
- Working Phone: http://pastebin.com/q3pHcwna
- Not working phone: http://pastebin.com/iiCHPMmn
2012/8/20 Rusty Newton rnew...@digium.com
On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
Hi
I've got a little issue
Hi All,
I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in
routing calls to upstream carrier via SIP trunks out. I spent a lot of time
in the lab testing 1.8 which included heavily testing DTMF with no issues
that came up. It all just seemed to work fine. But then
I had similar problems with 1.8.6 and polycom phones intermittently having
DTMF issues. I updated to 1.8.7 and things cleared up. I went through the
release notes at the time, but don't recall which commit made me decide to
give it a try.
Rgds,
Jared
On Wed, Nov 9, 2011 at 7:03 PM, JR Richardson
Hi,
I have a requirement where the DTMF entered by a member in konference is
passed on to the other members.
But the DTMF is not being recognized, when checked the events from manager
API, I do see DTMF event being passed, but some how it is not passed to
other members.
This tells me - may be
I am using H.323 to create a trunk between Asterisk and Avaya IP Office system.
Everything is working correctly, Asterisk
can call Avaya and vise versa. Now I create a conference room with a
user pin in Asterisk. Avaya can call into the conference room, but can
enter the pin number. The error
This is the setting i am using for Avaya CM to Aseterisk. (and pinf code
is working when dialing from Avaya to Asterisk
conference)sip:/etc/asterisk# cat ooh323.conf
[general]
bindaddr=213.88.138.183
port=5088
context=inputinterior.se
dtmfmode=rfc2833
;h323id=may day
;callerid=may day
Beside the port number and the alaw, the only difference is the dtmf. I added
this into my ooh323.conf and it still didn't work.
dtmfcodec=127
dtmfmode=rfc2833
I also tried: dtmfmode=h245signal
This is to an Avaya IP Office 500.
--
Make sure u have the correct DTMF over IP (or what it is named in IP
Office, thats the CM name) setting on the signal-group. In my case: DTMF
over IP: rtp-payload
On Wed, 20 Jan 2010 16:11:58 -0800 (PST), hin lee wrote: Beside the
port number and the alaw, the only difference is the dtmf. I
Hi all,
Kindly note that I got the below message when sending DTMF in RFC2833
through asterisk PBX...The DTMF is not going through
RTCP Read too short
I'm using G729 codec and asteriks Asterisk 1.4.21.2
Regards
___
-- Bandwidth and Colocation
Dear All,
I have the following scenario:
My customer dial a DID number and it'll be forwarded to my asterisk server
by the below trunk defined in sip.conf:
[sip_proxy1]
type=peer
context=stations
host=81.201.82.112
disallow=all
allow=g729
allow=alaw
allow=ulaw
dtmfmode=RFC2833
relaxdtmf=yes
went with the TDM Wildcard route
with analog lines. Good luck
Date: Thu, 16 Oct 2008 13:28:54 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue
Dear All,
I have the following scenario:
My customer dial a DID number and it'll be forwarded
Dear All,
What could be the problem if I try to send DTMF in RFC2833 format to my
asterisk server and it replies back with 603 error message?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September
Hello list,
i'm sure this is not a new issue, i'm having DTMF recognition issues with
TDM404.
I've already tried relaxdtmf=on/off and that did not do any good.
i was wondering if there is any where else in zaptel/zapata to play with and
have it fine tuning.
Or maybe this card is not handeling
Thanks maka and sorry I just saw your email (too many in my account). I set
relaxdtmf=yes and going to try it. Thanks again.
Larry
--
Message: 5
Date: Mon, 5 Sep 2005 12:48:28 +0300
From: maka [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DTMF issue on IVR
Hi All,
I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with
RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt
and asks caller to dial four-digit extension. Caller has to dial slowly,
otherwise, Asterisk cannot recognize the extension number. I
hya,
try using relaxdtmf=yes in zapata.conf and see if that solves it.
checkout these recent postings as well:
http://lists.digium.com/pipermail/asterisk-users/2005-August/122737.html
http://lists.digium.com/pipermail/asterisk-users/2005-August/122656.html
cheersOn 9/5/05, larry lin [EMAIL
On Fri, 2004-07-16 at 18:45, Andrew Yager wrote:
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds
I'm getting down to the last of my * issues ...
After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the other end still gets the digits
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes
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