Inability to break dial tone is a common problem encountered in a VoIP
network. In this scenario, the calling party is unable to pass the dual tone
multifrequency (DTMF) tones or digits to the terminating device. This, in turn,
does not let callers dial the desired extension or interact with the device
that needs DTMF tones (such as voice mail or interactive voice response [IVR]
applications). This problem could be caused by any of these issues:
DTMF tones are not passed.
DTMF tones are not understood.
DTMF tones are passed but are not understood due to
Other signaling and cabling issues.
This document addresses the most common problems and solutions.
There are no specific requirements for this document.
This document is not restricted to specific software or hardware
The information in this document was created from the devices in a
specific lab environment. All of the devices used in this document started with
a cleared (default) configuration. If your network is live, make sure that you
understand the potential impact of any command.
For more information on document conventions, refer to the
Technical Tips Conventions.
The router puts a seizure on the local PBX, but the dial tone remains
while the user is dialing.
Ensure that the dial-type is set as dtmf
on both the router and the PBX, as shown in the next sample output. Because the
Foreign Exchange Station (FXS) port does not pass on digits, this setting is
not available on an FXS port. However, this setting can be
changed on Foreign Exchange Office (FXO) ports and on receive and transmit (Ear
and Mouth [E & M]) ports.
Router(config-voiceport)# dial-type ?
dtmf touch-tone dialer
mf mf-tone dialer
pulse pulse dialer
In case of E & M, issue a show call active voice
brief command to ensure that you are receiving the answer
supervision from the PBX. The status of the call should be
active, if you have received answer supervision.
If the Telephony leg is still in the connecting
state, then the router will not completely close the audio path. If this is the
case, then you should contact the PBX vendor and ask them to provide answer
A workaround to this problem is to try to change the signaling on the
router to immediate (see the next sample output)
and then issue the auto cut-through command under
the voice port. The router can then bring the call up to
active state and cut through the audio.
Router(config-voiceport)# signal ?
delay-dial delay before dialing
immediate start immediately
wink-start start upon wink
Voice-port configuration commands:
auto-cut-through E & M auto cut-through without answer signal
Note: The signaling should match between the router and the PBX. Otherwise,
calls in one direction might not work.
In the case of analog E&M, ensure that all cabling is installed
correctly as described in
and Troubleshooting Analog E & M Interface Types and Wiring
Arrangements. Correct installation ensures that both transmit and
receive audio paths are mapped correctly. Incorrect installation can cause
audio paths not to establish properly and, therefore, the digits will not pass
correctly between the two connected devices. The desired extension is reached,
but the terminal device does not understand the tones when they are
In the case of a VoIP call from an originating gateway (OGW) to a
terminating gateway (TGW), terminating the call to a Telephony device might not
be understood. When you are passing DTMF tones through a compressed VoIP audio
path, some or part of the dual tones could become slightly distorted because
digital signal processor (DSP) codecs are designed to interpret human speech,
not machine tones. Usually, such distortion does not occur with earlier
compression codecs, such as G.723 or G.711, but later compression codecs can
cause distortion of in-band tones. Cisco IOS® Software Release 12.0(5)T allows
the DTMF tones to be passed out-of-band between VoIP gateways via three
different techniques. All of these techniques use the H.245 capabilities
exchange (part of H.323v2) to signal to the remote VoIP gateway that a DTMF
tone has been received and that the remote VoIP gateway should regenerate
Issue the dtmf-relay command under the VoIP
dial-peer on both sides. There are three different types of DTMF relays that
can be configured:
Router(config)# dial-peer voice xxx voip
Router(config-dial-peer)# dtmf-relay ?
cisco-rtp Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal DTMF Relay via H245 Signal IE
Try a different setting for the dtmf-relay
command. The cisco-rtp setting is proprietary to
Cisco and is available prior to Cisco IOS Software Release 12.0(5)T. The other
two settings follow the H.323v2 standards.
For Media Gateway Control Protocol (MGCP) networks, refer to
Fax (T.38) and DTMF Relay.
For session initiation protocol (SIP) networks, refer to
Multifrequency Relay for SIP Calls Using Named Telephone Events.
The sent in-band tones might be distorted because of the configuration
of the voice ports.
The tones sent across the network might have a signal strength that is
too low or too high. You can adjust the input
gain and output attenuation of
the signal to change the signal strength. The configuration is found under the
Router(config-voiceport)# input gain ?
<-6 - 14> gain in db
Router(config-voiceport)# output attenuation ?
<-6 - 14> attenuation in db
You can increase or decrease the signal at input. The exact value
varies from vendor to vendor (the Telco). Normally this is +7. However, you can
always try to increase or decrease by one until it reaches optimum stage. If
the values of these parameters are set too low or too high, you might have
problems. Adjust the values. The default values are 0 for both settings.
In addition to the previous issues, one-way audio can also contribute
to this type of problem. When there is one-way audio, the digits sent across do
not reach the intended destination. A common way to establish audio paths in
both directions is to issue the voice rtp send-recv
command on both routers. For more information to troubleshoot one-way audio,
One Way Voice Issues.
If none of these solutions resolve your problem, contact the
Cisco Technical Support.