Displaying 20 results from an estimated 20000 matches similar to: "Getting DTMF to work with SIP?"
2005 May 12
0
Asterisk, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.
Here is a rough diagram of my setup:
Asterisk |
server | NAT <------------ Libretel
| router
|
Note that there are NO SIP
2003 May 23
4
SIP and DTMF
Hello,
I am fairly new to asterisk. I am currently using asterisk as a
more convenient sip side voicemail system.
My problem:
I have cisco 7960 phones whose out of band dtmf tones
are recognized properly(when dtmfmode=rfc2833) by asterisk but whose
in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For
example 7999 comes out as 799999, 4242 comes out as 442422 ... etc
I
2006 Jan 23
0
DTMF not working on overseas cellphone calls
I thought I sent this earlier this week, but I didn't see it. If I
missed it, I apologize for the resend.
We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On
incoming calls from cellphones located overseas, DTMF is not recognized
- we have many single-digit choices in our menu so the problem isn't
that some digits aren't working, it's not listening at
2013 May 24
1
Asterisk 11 dtmf not recognised
Hi
I have a dialplan as per the following,
extensions.conf
[avgtest]
exten = 100,n,Playback(avgtest/message1)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,,5,3) ; Attempts for 5 times with 3 seconds of
timeout
exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
exten = 100,n,Hangup() ; Didn't go to pin-accepted, so play badPIN and
hangup
2006 Jan 19
0
DTMF not recognized on overseas call from cellphone
We have PSTN lines connected to FXO lines of a TDM400B. I just got a
complaint that overseas callers who are using cellphones sometimes find
that DTMF digits aren't working - they press digits and the menu goes on
as if they hadn't pressed anything. Since it sometimes works, and other
IVRs work over the same cellphones, it's not that the cellphone isn't
sending the digits.
2005 Jan 05
3
Sending DTMF to PSTN problem with SIP
Dear All ~
I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN & try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain seems to go wild (high), and the
DTMF tones are not recognized at the other end.
I tried setting the
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
2007 Apr 09
2
DTMF auto detection bug?
Hi,
it seems that there is a bug in asterisk's dtmf mode autodetection.
Assume following sip.conf:
[sipprovider]
disallow=all
allow=g726
dtmfmode=auto
DTMF does not work. It seems rfc2833 mode is chosen despite it being
obvious that this cannot work!
The following configuration is necessary to get DTMF to work: dtmfmode=info
In my opinion, this behaviour is counter-intuitive. I am using
2005 Mar 26
1
DTMF tones not working
I have Polycom ip-300 phones that worked yesterday but dont seem to work
today (at least dtmf signalling once connected to the asterisk box)
The current configuration is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = test
srvlookup = yes
dtmf = inband
allow = all
dtmfmode=inband
progressinband=no
disallow=all
allow=ulaw
pedantic=no
[202]
type=user
secret=xxxx
context=test
mailbox=202
2003 Dec 21
1
SJphone, Asterisk and DTMF tones ...
Hi,
I am using SJPhone here for testing ivr with Asterisk. I haven't seen any
problem here yet.
I have tried different things for that and finally got it working. I am not
an expert to explain more about that, but here is the general section form
my sip.conf. dont know whether it will help...
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ;
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
2009 Mar 17
0
DTMF troubles
I've been using one of the popular asterisk ISO distributions for a
couple of years and DTMF had always worked.
I recently switched to another asterisk ISO distribution, and outbound
DTMF is no longer working.
After doing a bit of digging, I noticed that the new distribution
wasn't setting any sort of dtmfmode at all, anywhere. The old
distribution had rfc2833 set in sip.conf
2003 Dec 30
1
SIP + DTMF problem
I am having a
problem interacting
with a remote IVR
system when the
outbound call is
going via SIP. The
only way that I have
been able to get a
response from the
IVR is to set
dtmfmode=info in
sip.conf.
Unfortunately that
doesn't quite fix
the problem because
it will still only
accept DTMF input
once the voice
response has
finished on the IVR.
If I try and press
anything while the
IVR is
2005 May 23
0
How to detect DTMF and change if needed
I have done some searching and not sure this is even possible, but
here it goes...
**Scenario**
Let's say you have an asterisk server that you use to connect to a SIP
provider that you push your PSTN-bound calls to using g711 and
out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set
to also use out-of-band DTMF. For the most part, everything works
great.
However, a few
2004 Jan 25
2
Incoming SIP matching
Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to
have dtmfmode=rfc2833. However, incoming FWD calls from the dialup
access numbers (such as libretel) need to have dtmfmode=inband. To
solve this problem, I created a second FWD account and configured
sip.conf as follows, in order to match the incoming number to the proper
dtmfmode:
[fwd-rfc]
type=friend
secret=*****
2004 Apr 26
3
dtmf tone clamping in calls to external ivr
Hello,
I'm having trouble working out how to send DTMF tones to an external
IVR. My system has an analog phone connected to a TDM400P card, a SIP
software phone (Zultys LIPZ4) and is connected to a BRI in Australia
with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched
with the ISDN audio patch from Traverse (which allows the card to do
voice).
DTMF works fine between
2007 Oct 04
2
Voicemail/dtmf not working?
Hi,
I am setting up an asterisk server for testing purposes and cannot get
voicemail to work at all.
My host OS is Linux From Scratch 6.3 and the asterisk software versions
I built are zaptel-1.4.5.1 and asterisk-1.4.12.
I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk
server and client phone are on different computers but are on the same
LAN, i.e. no NAT.
I have an
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
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
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