DTMF problems

Asterisk Asterisk Support
1

Hi,

I made a simple IVR with asterisk. There is a B410P card in the PC and it’s connected to the PSTN via 2 BRI interfaces.

If i call the IVR extension from a cell phone or a land line phone everything works perfectly.

However, if i use a VoIP phone (not connected to the asterisk server in any way) with a VoipDiscount (or any other VoIP provider for that matter) account and make a call to the *, then i have some trouble. Asterisk misses some of the keys I press on the phone, and only recognizes them sometimes, making navigation in the IVR impossible.

Again, if i use a regular phone to call the *, everything works.

I guess it has something to do with the way the voip provider sends the dtmf signals.

My question is, how can i adjust the settings for my B410P card? I looked in misdn.conf and misdn-init.conf, but could not find anything relevant to my problem.

2

are you using the DTMF=RFC2833 setting?

3

In which config file do I specify that?

4

hi mcduglas,

      You want write this information in sip.conf at the bottom which is situated in '/etc/asterisk' directory.

[101]
callerid=101 <101>
canreinvite=no
dtmfmode=rfc2833
host=dynamic
nat=never
port=5061
qualify=no
record_in=Adhoc
record_out=Adhoc
secret=101
type=friend
context=sip
username=101

Note:

  1. If u r using u’r softphone outside an network where asterisk runing then you need
    following changes
    i) make nat=yes
    ii) make qualify=yes

  2. and yes very important thing is context, write down that context which u’t mentioned in
    extension.conf. If u not used an context i.e. you used default context then write down
    context=default or otherwise write down context=XXX (defined in extension.conf)

  3. and make sure that you don’t have any firewall problem otherwise you don’t receive an
    rtp packet which is responsible to transfer or receive dtmf tone (multimedia)

cheers,
deepen

5

I don’t understand why do i need SIP configuration on my Asterisk server. I have no SIP devices, or SIP providers connected to this * box.

SIP is only involved at the client’s site, where i have no possible access to configure anything.

Here is the setup:

bad: ----- ----- -----
good: ----- -----

So, there is no SIP on my side, only a Digium B410P card with 2 BRI connected to it.