ASL3 Sip extension not working

1

So I setup a ASL3 extension, and I am able to successfully connect to the extension, I can dial into the node, but I cannot execute dtmf commands and it drops the call after 30 seconds.

In my sip.conf I have:
[201]
type=friend
host=dynamic
username=201
secret=
dtmfmode=rfc2833
context=sip-phones
callerid=“Softphone” <201>

I am using port 5160 for Sip.

In extensions.conf I have:

[sip-phones]
exten => 201,1,Dial(PJSIP/${EXTEN},60,rT)

exten => 62038,1,Ringing
exten => 62038,n,Answer(3000)
exten => 62038 ,n,Set(NODENUM=${CALLERID(number)})
exten => 62038,n,Playback(extension)
exten => 62038,n,SayDigits(62038)
exten => 62038,n,Playback(connected)
exten => 62038,n,Playback(rpt/node)
exten => 62038},n,SayDigits(${EXTEN})
exten => 62038,n,rpt(${EXTEN}|P)
exten => 62038,n,Hangup
2

1st I might suggest using 4 digit SIP extensions.

I see no reason to set the var NODENUM

Shouldn’t this be.something like…

exten => 62038,n,Set(CALLERID(num)=${NODENUM})

Not sure what the outcome was suppose to achieve.
One, I would set it before answering it, even ringing it.
But the normal CID would be the dialing device extension I think. Provided it was set at extension stanza. Not sure why to change it.
Perhaps you have reasons.

When dialing from the node (phone patch), you can set the CID name and number like this in rpt.conf for each node stanza
[29285]
callerid = “KB8JNM Repeater” <29285>
That will show in whatever SIP phone you dial from the node unless you change it.