DTMF not being recognized by remote IVR

Asterisk Asterisk Support
1

Hello,

I’m using asterisk 1.4 on CentOS 5, VoicePulse trunk and Polycom SoundPoint IP 650 phones. Everything is working fine except that when we call certain numbers that have IVR, some IVRs don’t recognize the digits we dial. You can press 1 quickly or slowly and nothing happens on the other side. I tride to change dtmfmode=inband but this didn’t help. Any idea how to fix this problem? No Zaptel involved here, just SIP and IAX2.

2

Hi Majamar,

  Don't change dtmfmode to inband but keep dtmfmode=rfc2833 only. And yes please check you forward  rtp and sip ports open in firewall. You also try out by disabling an firewall once. use" rtp debug" command and watch that you gets both "send" and "got"  packet.

Please write down more detail if your problem not solved so we try to solve your problem.

Sorry for delay.

Thanks,
Deepen

3

Thanks for your reply. I can’t diable the firewall in my Cisco IOS router. However, all ports are forwarded to the Asterisk server, and according to Cisco, the RTP ports are mapped automatically to the Asterisk server through the SIP algorithm. How does the rtp debug command work? I read the books and cocs, but nothing about what rtp debug ip does. Does it save a file somewhere? I have -rvvvvvvvv and still don’t see anything in the console after I enable debug.

4

Hi,

give rtp debug command on asterisk console. And just establish a call. You show there are continuous sent and got rtp packet on asterisk console.
And I am sure you only gets an send packet in console not got. If you see this behavior on console then there is 99% firewall problem.

Regards,
Deepen

5

Hi,

give rtp debug command on asterisk console. And just establish a call. You will see there are continuous sent and got rtp packet on asterisk console after call established.
And I am sure you only gets an send packet in console not got. If you see this behavior on console then there is 99% firewall problem.

Regards,
Deepen

6

What codec are you using?

7

Thank for the tip. rtp debug is helpful. my related problem in other posting could be a firewall/nat issue.

regards

8

Hi,

is rtp debug only for sip based phones? even the locally connected phones are not recognized in rtp debug. So, I still have the digits recognition problem with my pstn.

regards