[Asterisk-Users] dtmf problem

[Bartosz Wegrzyn - asterisk]

hi everyone,

I do have 2 lines with broadvoice.
>From 2 days on one line my dtmf tones are not passed to asterisk server.
It siply goes through the extensions routine acting link it did not
receive any tone. Could it be problem with my config???

It looks like this:(it worked for last 1.5 year)
num2 is ok, but num1 is not working.

Any ideas before I call support which is always a problem.

[general]
externip=lexon.ws
bindaddr = 192.168.1.251
port=5060
localnet=192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
register => number1:pass1 at sip.broadvoice.com
register => number2:pass2 at sip.broadvoice.com/2000
tos=0x18
srvlookup=yes
nat=never
insecure=yes

[sip.broadvoice.com]
type=peer
username=num1
fromuser=num1
authuser=num1
secret=pass1
host=sip.broadvoice.com
context=sip
fromdomain=sip.broadvoice.com
canreinvite=no
nat=never
dtmfmode=inband


[sip.broadvoice.com.home]
type=peer
username=num2
fromuser=num2
authuser=num3
secret=pass2
host=sip.broadvoice.com
context=sip
fromdomain=sip.broadvoice.com
canreinvite=no
nat=never
dtmfmode=inband

[broadvoice-incoming]
type=peer
host=147.135.8.128
context=from-broadvoice
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
nat=never

[broadvoice-incoming2]
type=peer
host=147.135.0.128
context=from-broadvoice
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
nat=never

[broadvoice-incoming3]
type=peer
host=147.135.4.128
context=from-broadvoice
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
nat=never

thx

Bart Wegrzyn