DTMF isn't working

Asterisk Asterisk Support
1

Hi,

I am running asterisk 1.8.2 on CentOS 5.
When I make a call from Voxeo to the cell phone, both DTMF and speech recognition are working.
But when I make a call from the cell phone to Voxeo, I can hear prompts, but DTMF isn’t working.
What am I missing here?

Here’s my environment. No additional telephony hardware is involved.
a cell phone <–>SIP Provider<–>Asterisk(NATed)<–>Voxeo (as a soft-phone)

Here’s my sip.conf:

[code]pedantic=no
register => 1231231234@sip.provider.com:pass:1231231234@sip.provider.com
localnet=192.168.10.0/255.255.255.0
externaddr=123.123.123.111:5060
nat = comedia

[sip.provider.com]
type=peer
context=inbound
host=sip.provider.com
fromdomain=sip.provider.com
fromuser=1231231234
secret=pass
username=1231231234
insecure=invite
authname=1231231234
dtmfmode=rfc2833
dtmf=inband
canreinvite=yes
directmedia=yes

[voxeo]
type=friend
host=dynamic
secret=pass2
context=outbound
dtmfmode=info
dtmf=info
deny=0.0.0.0/0
permit=192.168.10.0/255.255.255.0[/code]

Here’s my extensions.conf:

[code][inbound]
exten => _NXXNXXXXXX, 1, dial(SIP/voxeo,30)

[outbound]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.provider.com, 30)
[/code]

2

I made a few changes to sip.conf, made a few calls, and reviewed RTP traffic generated in the following cases.
It looks like DTMF events with payload type being telephone-event (101) are working whereas those with payload type being DynamicRTP-Type-101 (101) are not. The payload type seems to have something to do with dtmfmode and dtmf property in sip.conf. How can I fix this problem? Thanks.

  • case 1 (I made a call from Voxeo to my cell phone with the same sip.conf shown in my first post. DTMF doesn’t work here.)
  • case 2 (I made a call from a cell phone to Voxeo.)
    With the following changes to sip.conf, DTMF works.
[sip.provider.com]
type=peer
context=inbound
host=sip.provider.com
fromdomain=sip.provider.com
fromuser=1231231234
secret=pass
username=1231231234
insecure=invite
authname=1231231234
dtmfmode=rfc2833
dtmf=inband
canreinvite=yes
directmedia=yes

[voxeo]
type=friend
host=dynamic
secret=pass2
context=outbound
dtmfmode=rfc2833
dtmf=inband
deny=0.0.0.0/0          
permit=192.168.10.0/255.255.255.0
3

Which codec? In band DTMF will only work reliably for mulaw and alaw.

4

In my previous tests, I didn’t specify allow or disallow properties. So by default, allow =all was used.

I tried different combinations of dtmfmode and dtmf values, and setting dtmfmode=auto and dtmf=auto for both voxeo and sip.provider.com gives me the best results, in which dtmf won’t work only if I initiate an outbound call from voxeo to my cell phone. DTMF is supposed to work regardless of which endpoint initiates a call, right? But why am I running into this problem?

5

What codec(s) do you want to be able to use in your system? Once we know that, we can better recommend optimal settings/troubleshoot those settings if they aren’t functioning correctly.

6

for now I would like to use g711 PCMU.

7

Hello,
I have problem with dtmf on SIP freecall.com

Asterisk 1.4.21.2~dfsg-3+lenny2

in SIP.config i set:

dtmfmode=rfc2833 dtmf=auto rfc2833compensate=yes

in extensions.conf i use SendDTMF(12345,250)

and Asterisk system call to my phone, and I not hear DTMF code.

in debug sip i have:

What can I do to make this work?