[asterisk-users] DTMF transmission problem

Noah Engelberth
Thu, 02 Aug 2012 12:45:28 -0000

I am having difficulties with customer-bound DTMF being very short & clipped
off (and basically unusable, as systems on the customer side aren't
recognizing the DTMF digits, and I can barely tell that DTMF is there when I
listen on a handset).

My system set up as follows:

PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE

Asterisk is running Asterisk 10.4.0 on a CentOS 6.2 VM residing on a CentOS 6.3
KVM host.  Asterisk has one network interface connected to the Metaswitch
without NAT to place/receive calls from the PSTN, and a separate interface to
connect to CPE equipment.  SIP and IAX are bound to both interfaces.  Vocal call
quality is fine, DTMF is fine from the customer to the PSTN, but DTMF from the
PSTN to the customer isn't.  Asterisk is set to remain in the media path on
all calls.  The customer facing IP address on the Asterisk server is private and
is being 1:1 NATed through a MikroTik RB 1100 to a public address that the
customers are then connecting to.  I have also placed test calls with the
"customer equipment" inside the same LAN as the Asterisk server's
customer facing IP address (no NAT) with precisely the same symptoms.  The same
symptoms persist whether the PSTN or the CPE initiate the call.

My example configs are as follows:

SIP -
[general]
limitonpeer=yes
notifyringing=yes
notifyhold=yes
allowsubscribe=yes
disallow=all
allow=g722
allow=ulaw
allow=gsm
allowoverlap=no
callevents=yes
allowguest=no
directmedia=no

bindport=bind_here
bindaddr=to_this_address
srvlookup=yes
maxexpiry=7200
defaultexpiry=3600

[authentication]
[test-voice]
type=friend
host=dynamic
secret=not_my_secret
context=users
disallow=all
allow=ulaw
nat=yes
directmedia=no
qualify=yes
trunk=no

IAX2 -
[general]
bindport=bind_here
bindaddr=to_this_address
delayreject=yes
disallow=all
allow=g722
allow=ulaw
allow=gsm
jitterbuffer=no
encryption=yes

[test-fax1]
type=friend
host=dynamic
username=test-fax1
secret=not_my_secret
context=users
disallow=all
allow=ulaw
qualify=yes
trunk=no
requirecalltoken=no


SIP peers are Zhone ZNID-2xxx series ONTs.  IAX peers are ATCOM AG198 ATA
gateways, either behind the ONTs (but on the same voice VLAN the ONTs use to
talk to Asterisk) or on my Asterisk server's local network.  The voice VLAN
is a different subnet than Asterisk is on, but no NAT exists between the
subnets.

Thank you,

Noah Engelberth
System Administration
MetaLINK Technologies

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