READ cmd not grabbing digits

exten => 900,1,Read(NUM,4,3)
exten => 900,n,SayNumber(${NUM})

In the cli:
– Executing [900@internal:1] Read(“SIP/102-083409c0”, “NUM||4|3”) in new stack
– Accepting a maximum of 4 digits.
– User entered nothing.
– Executing [900@internal:2] SayNumber(“SIP/102-083409c0”, “”) in new stack
[Jan 6 14:59:12] WARNING[3674]: pbx.c:6050 pbx_builtin_saynumber: SayNumber requires an argument (number)
== Spawn extension (internal, 900, 2) exited non-zero on ‘SIP/102-083409c0’

Im not sure whats going on. I tried it from Xlite and a Polycom.
I used this same config file on AsteriskNOW then i formatted and put on Centos5 and newest Asterisk and it wont work. The xlite settings are the same, along with the Polycom, along with a Grandstream… I havent changed anything on these phones…

Anyone have a clue on what I should check?

Hi Try

exten => 900,1,Read(NUM,,4,3) exten => 900,n,SayNumber(${NUM})

as the 3 is in the option postition if you want a timeout its

[quote]exten => 900,1,Read(NUM,4,3) ; 3 second timeout
exten => 900,1,Read(NUM,4,3) ; 3 attempts

exten => 900,n,SayNumber(${NUM}) [/quote]

also check DTMF is working.

Ian

exten => 900,1,Read(NUM,,4,3) exten => 900,n,SayNumber(${NUM})

Yea thats what I posted in my first post…

I tried all 3 varitations you posted anyways just for the heck of it, no worky.

I did this test and called it and hit a bunch of digits:
exten => 905,1,Read(test)
exten => 905,n,NoOp(DTMF Received: ${test}
exten => 905,n,Hangup

-- Executing [905@internal:1] Read("SIP/102-09f653e0", "test") in new stack
-- User entered nothing.
-- Executing [905@internal:2] NoOp("SIP/102-09f653e0", "DTMF Received: ") in new stack
-- Executing [905@internal:3] Hangup("SIP/102-09f653e0", "") in new stack

== Spawn extension (internal, 905, 3) exited non-zero on ‘SIP/102-09f653e0’

sip.conf:
[general]
context=internal
realm=test.com
videosupport=yes
recordhistory=yes
musicclass=default
language=en
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
trustrpid=no
promiscredir=no
usereqphone=no
dtmfmode=rfc2833
compactheaders=yes
tos=lowdelay
pedantic=no
nat=no
defaultexpiry=360
maxexpiry=3600
disallow=all
allow=gsm
allow=ulaw
allow=alway
srvlookup=yes
bindport=5060

[authentication]

in-reg
type=peer
canreinvite=no
insecure=very
qualify=yes

users-common
type=friend
host=dynamic
canreinvite=no
qualify=yes
nat=yes

100
username=Matt
callerid=Matt<100>
secret=wishUknew
mailbox=100

Im going to get my digium card hooked back up tomorrow night and test DTMF through the PSTN…

Hi

I note that on the extension you say nat=yes but I dont see you defining the localnet at all or th externip.

Ian

[quote=“ianplain”]Hi

I note that on the extension you say nat=yes but I dont see you defining the localnet at all or th externip.

Ian[/quote]

So I would just add the following?

[general]
externip = 70.123.123.123 ;myWANip of DSL Modem
localnet = 192.168.3.0/255.255.255.0 ;Asterisk network(asterisk is 192.168.3.32)

Thank you

ps…what is this good for? ive used soft sip phones outside my wan with the current settings and it works fine. What does these settings do differently. THanks

i just got the digium card going, i can send DTMF through it no problem…

actually I seem to have sip dtmf mysterisly working now too…
if you could still help me on the post 2 up that would be appreciated too!

Changing the ip address did the trick. Its’ no mystery I suppose.