Displaying 20 results from an estimated 10000 matches similar to: "DTMF with duration = 0"
2009 Apr 13
3
duration of rfc2833 generated dtmf
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like when RFC2833 is used, it actually generates
rtp packets of some sort, so I have no idea how to increase that
duration.
Any assistance would be appreciated.
--
Your
2008 Dec 19
1
Increase DTMF Tone Duration
Hi,
We are running 1.4.22 and have been experiencing problems with certain
IVRs and DTMF Tone duration. We would like to be able to increase DTMF
Tone duration by 50 to 100ms over what the user is pressing on his
phone. We have a PRI test circuit and an analyer in between to measure
tone duration.
We have tried setting chan_dahdi.conf parameter 'toneduration', but that
does not do
2011 Jul 05
0
DTMF between sip trunks and PRIs
Hi,
I'm looking for some advice on how to solve DTMF issues.
I have 2 boxes, one which is the connection to the PSTN (PSTN) through
PRIs and SIP trunks, and a second (PBX) which has UAs registered to
it.
We have a customer that has an existing pbx that we trunk analog lines
to using a GXW-4008.
The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF.
The issue I'm
2011 Nov 10
0
DTMF issue with 1.8.6.0 and SIP Trunks [WORKING]
> 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 again you can?t
> reproduce every real work scenario in the lab.
>
>
>
> I?m
2007 Jun 20
1
DTMF doesn't work between Asterisk and Cisco SIP Proxy
Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service
The Call can be established,and I can hear from x-lite the prompt of
the conference,but when I input any digits,nothing happened,the
conference service did not recognize my input.At the same time,in
2006 Feb 28
2
Sipura SPA-3000 and PSTN dtmf
Greetings,
What is the recommended settings for using SPA-3000's FXO port for
dialing out to PSTN in regard of the DTMF?
The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports
registered to the Asterisk box with unique username/passwords.
The inbound PSTN DTMF works excellently, e.g. people calling from PSTN
into the * box are able to pick IVR items with DTMF reliably.
The
2011 Feb 18
2
DTMF and Snom
Hello list,
I'm having some troubles with DTMF tones. When pressing numbers on a
Snom phone, the DTMF-signal takes too long.
I have the following in sip.conf :
dtmfmode = rfc2833
which works well for Grandstream, Yealink and Cisco phones. But not for
Snom.
Snom support tells me I should use SIP info.
Is it possible to have something like this :
dtmfmode = rfc2833, info
??
Because
2004 Dec 21
0
SIP dtmf=rfc2833 not working
We are testing some DTMF-driven applications over VOIP (legacy systems
which use fast pulses of standard DTMF tones).
The applications work fine when Digium IAXy's are used - no loss or
garbling of DTMF tones.
However, when we use SIP modems (such as Sipura 1000's), the DTMF tones
are frequently uninterpretable and our applications have to ask for
retries.
I am under the impression that
2003 Nov 19
0
SIP/IAX2 DTMF
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
When making a call like the one below, I get double DTMF tones on the PSTN
side. DTMF tones sent from the PSTN arrives squelched on the SIP side.
SIP > Asterisk2 > IAX2 > Asterisk1 > ZAP > PSTN
SIP has been configured to use rfc2833 on both the SIP endpoint and the
Asterisk. SIP endpoint also suggests a payload value of 101.
2009 Aug 25
0
DTMF duplicated when Waitexten
Hello,
I have a problem of DTMF duplication.
I receive call from my provider with SIP protocol. These calls pass
through an interactive voice menu, using the application Waitexten to
enter a client code. The menu works fine, but sometimes I have DTMF
duplication that prevent proper code entry. All DTMF come twice.
my sip.conf
-----------
[general]
context=default
allowguest=no
2005 Jul 21
0
re: DTMF woes, continued
hello all,
I have a DID from nufone, transported via SIP to my * box, and even
though i'm using rfc2833 DTMF i'm still getting double digits and all
sorts of other stuff...
sip.conf is as follows:
[general]
port = 5070 ; Port to bind to
disallow=all ; Disallow all codecs
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
dtmfmode=rfc2833
register =>
2009 Feb 16
1
DTMF not completely muted
Hi all,
When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms
and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips,
at the end of the recording.
I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards:
a TE420 w/Octasic and pri_net
2005 May 14
0
Transferring a call, IAX2->SIP, DTMF/RFC2833 doesn't work?
We are using Asterisk 1.0.7. We have this scenario:
IAX2 user comes in to Asterisk, dials an extension, and transfers to a
SIP user.
The dial command is simple, looks like this:
exten => 300,1,Dial(SIP/300)
Extension 300 is a legacy PBX device operated by touchtones. The user
(coming in over IAX2) is trying to drive this PBX using touchtones. But
the trouble is, by the time the touchtones
2008 Dec 24
0
DTMF Problems
First of all Merry Christmas.
Second, my first problem with my provider not staying registered with
our server was my fault. We moved our server room and I restarted the
test system and the production system causing them to ping-pong back and
forth registering with our provider causing random problems, they are
both set to register with the same account right now. I shut Asterisk
down on
2006 Mar 16
1
RFC 2833 and SIP? DTMF? What am I not getting?
Hi again,
I am trying to get my DTMF to use RFC 2833 rather then inband, so that
I can utilize lower bandwidth codecs through my FXO.
After much tinkering I was able to get my gateway (wellgate 3701A)
configured to a point where I have some success, but no real joy.
I have configured the RTP Payload type (or RFC2833 Payload type) to
101. I don't have a clue what this means, but I took
2010 Jun 29
1
Asterisk 1.6 (and 1.4) DTMF problems using RFC2833
We are experiencing intermittent DTMF problems here, with the following
setup:
ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF).
I am running Ubuntu server 10.04, but Asterisk is compiled by us and not
installed from the software repository. Essentially, DTMF works for some
time, but at some point it simply stops and the point at which it stops
appears to be random.
Using RTP debug, I
2005 Jan 18
0
DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP
I am having a problem trying to do inband DTMF passthru via asterisk.
My setup:
PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2)
Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly)
12/10/04 and 01/17/05 (no difference)
CAC ABII-T100P to/from analog lines to/from asterisk
BTW, I have used a ABI and it works just like the ABII with asterisk.
What I am seeing is:
I make a call from a
2009 Jan 29
0
[asterisk-dev] DTMF queuing
[moving to asterisk-users by request]
On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jtodd at digium.com> wrote:
>
> On Jan 26, 2009, at 7:38 PM, James Lamanna wrote:
>
>>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote:
>>>
>>>> Hi,
>>>> Is it just me, or does DTMF queuing not work properly?
>>>> I'm consistently faced with
2007 Jan 10
0
DTMF on Snom
Hi all,
I have problem using DTMF on Snom Phones (300, 320 and 360)
I read they use in preference out-of-band DTMF , and if the remote system
does not support it they default back to inband.
I would like to use DTMF as out of band , and I defined
dtmfmode=rfc2833
in the peer configuration.
Nope, I am no able to access any ouside services using DTMF;
Another kind of phones, ATCOM AT320, can be
2008 Dec 29
1
DTMF does not work
I got no resonses to this and some funny bounces so I'm trying again.
First of all Merry Christmas.
Second, my first problem with my provider not staying registered with
our server was my fault. We moved our server room and I restarted the
test system and the production system causing them to ping-pong back and
forth registering with our provider causing random problems, they are
both