Impossible to authenticate to the server using dtmf from out

Asterisk Asterisk Support
1

Hi everyone,

I am using Asterisk 1.8.9 on a CentOS 6 OS (with a2billing 1.9.4). I 'm trying to configure a calling card access using a DID number. I can access to the IVR by dialing the DID number . There is no issue when i am using peer account (sip) configured in the asterisk system, but the issue is that when i am asked to enter my PIN code, DTMF tones aren’t recognized by the system when i am using a phone from outside ( mobile phone , gsm …). It would seem that something is missing in one of my config files . I need some help please .

2

is your did getting to your asterisk via g729?
it’s not recommended to send inband DTMF with g729
check your sip dtmf settings

3

Thank you nmirceac for your answer, i’m not using g729 actually. I 'm using the allaw codec. It would seem that the g729 codec is not installed by default in asterisk because it isn’t present in the /usr/lib/asterisk/modules directory.
Last thing : - can you tell me where can i check my dtmf setting elsewhere than in my /etc/asterisk/sip.conf
- is there something to download or to configure in asterisk to have a functional dtmf system ?

Here is my sip.conf configuration:

[general]
context = default ; Default context for incoming calls
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
videosupport = yes ; Enable video
disallow = all ; First disallow all codecs

;allow=g729 ;
;allow=g726

;allow = ulaw ; Allow codecs in order of preference
allow = alaw
;allow = gsm

allow = h263 ; H.263 is our video codec
allow = h263p ; H.263p is the enhanced video codec
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF.
rfc2833compensate=yes
canreinvite=no
nat=yes
qualify=yes
rtcachefriends=yes
relaxdtmf=yes

Thank you

4

There is nothing extra needed for DTMF. The only requirement is that dtmfmode in sip.conf is compatible with the settings on the peer (and that dtmfmode=inband is only used with simple codecs - mainly G.711 A and mu-law).

You will need to provide debug level information to diagnose this further.

5

Thank you david55 but even by replacing the dtmfmode value by inband that still doesn’t work. How could i install the g729 codec for free and configure it please ?

6

G.729 isn’t free. It’s patented intellectual property. G.729 can be used with Asterisk:

store.digium.com/productview.php … 8G729CODEC

7

G.729 won’t help you here, anyway; it forces you to use out of band “DTMF”.

8

Also, when you said you tried inband, I got the impression that you were stabbing in the dark. You need to find out how the peer is configured, and match that, rather than trying everything until something works.

9

Of course david55 , u 're really right. My provider accept rfc2833 and inband dtmfmode. the thing is that i tried

disallow =all
allow=g729
dtmfmode=rfc2833

and

disallow=all
allow=allaw
dtmfmode=inband

but nothing really good.

10

I would use alaw (1 L) and 2833.

I’d also get a wireshark trace and confirm that the DTMF is actually being sent.

11

Hello,

Seeing that Google ranks this thread high in the list of search results for this keyword: “dtmf authentication pin code”, I thought I share this Codeproject article that I just found this morning: http://www.codeproject.com/Articles/794853/How-to-create-an-IVR-based-telephone-client-gate-s

Quoting from the article: “This article presents how to develop a C# application using XML/HTTP/PHP and DTMF signalling that allows you to authenticate your customers through a phone call with the help of their User ID and PIN code.” The writer has used the prewritten VoIP components of Ozeki VoIP SDK.

After checking its website, I found this page about how to use this SDK with Asterisk: http://voip-sip-sdk.com/p_125-asterisk-pbx-setup-voip.html

I hope it can help you and even your readers as well! :wink: