ASL3 DTMF from SIP phone

AllStarLink
1

Hello all! I have a few ASL3 boxes and another box with a PBX on it. Recently realized that commands issued from a phone while on an active call are not acting upon rpt. Specifically, I am able to connect to any of my nodes as usual but should I dial *70 from a phone, I see the DTMF displayed on the ASL3 box but nothing happens. Dialing *70 from a radio works fine.

Not sure if maybe I am missing a new parameter in ASL3, I have been able to command the nodes real time ever since the ACID days, maybe before. Searched through existing posts and haven't found the answer. Any suggestions?

Thanks. 73

Kyle

K0KN

Olathe, KS

2

if you are running it thru IAX connection or over a VOIP connection, then you must issue the star99 before the command. ex: star99 star70 #. replace the word star with the asterisk " * ".

3

Hello. Thanks for the reply. I believe you're referring to regular "Phone" mode where *99 is dialed for PTT? I am using Vox mode so that my mute button essentially becomes PTT. I did try using phone mode yesterday in case a change was made that affected DTMF in vox mode.

My dialplan commands are as such:

exten => 1234,n,Rpt(${EXTEN},Pv)

4

The 1234 should match your allstar node number

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On Thu, Sep 24, 2026, 1:48 PM K0KN <[email protected]> wrote:

Someone replied to your post.

| K0KN
September 24 |

  • | - |

Hello. Thanks for the reply. I believe you're referring to regular "Phone" mode where *99 is dialed for PTT? I am using Vox mode so that my mute button essentially becomes PTT. I did try using phone mode yesterday in case a change was made that affected DTMF in vox mode.

My dialplan commands are as such:

exten => 1234,n,Rpt(${EXTEN},Pv)

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5

Kyle, you might want to review the 'RFC' 'MODE' of your phone.

One mode uses dtmf audio and others use digital signaling.

I use dtmf audio in most phones but 'speakerphone' mode can make that unreliable.

Perhaps test it with a sip-phone app on your pc.

6

Thanks Mike. I typically use out-of-band RFC2833. Asterisk doesn't have any issues decoding my DTMF per se, but it seems that app_rpt is not acting on the DTMF it receives? I suppose I could try in-band as a test.

7

I guess I was saying make sure the phone setting matches the sip config.

And again, speakerphone can monkey things up sometimes.

That's the only leads I have to explore outside of potential audio level issues.

Watch asl in foreground for dtmf decoding.