VOIP Provider no inbound / outbound DTMF

MST

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#1
Hello ALl,

I am trying to figure it out how to resolve problem with DTMF. The provider is Voip Innovation
Tunk:

Code:
type=peer
trustrpid=yes
dendrpid=yes
host=X.X.X.X  VOIP's provider IP
context=from-trunk
disallow=all
allow=rfc2833&ulaw&g711
dtmfmode=inband
canreinvite=no
insecure=very

USER CONTEXT: phone number

USER DETAILS:
canreinvite=no
context=from-trunk
fromuser=phone number
qualify=no
type=user

REGISTER STRING:

ph number/VOIP provider IP
All works except DTMF that are not recognized during IVR welcome message. The same thing outbound. All ports and access rules is open between Elastix IP and Provider IP.

RTP = 10000-10300 SIP_NAT is setup accordingly.

Thank You for your advice.

MST
 

dicko

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#2
allow=rfc2833&ulaw&g711
dtmfmode=inband

rfc2833 is not a codec so that line is wrong.
SIP usually works better with
dtmfmode=rfc2833 or (with some lame providers, auto)
 

MST

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#3
"rfc2833 is not a codec so that line is wrong" - ok I have fixed that


dtmfmode=rfc2833 does not work
dtmfmode=auto does not work
dtmfmode=inband does not work
 

dicko

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#4
then try:

A ) turn DTMF debugging on and try and fix it

or perhaps easier:

B ) Try another carrier.

dicko

p.s.

ulaw is g711(u) as alaw is g711(a) but g711 is not a valid codec either (maybe a little of that old RTFM'ing needed here? )
 

MST

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#5
console => notice,warning,error,verbose,dtmf

restarted asterisk

disallow=all
allow=ulaw
dtmfmode=inband

CLI did not show any DTMF

Can DTMF be up to RTP ?
 

tak.cassidy

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#6
Hi,
are you using sip phones or analouge phones through an ATA. If the later try checking the settings of what the ATA is sending.
Tony
 

dicko

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#7
you will better see those events in /var/log/asterisk/full
 

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