Syntax Difference

Hello everyone, i have a “beautiful” Grandstream HT 488 attached to Asterisk 1.4.0 "last CVS"
With release 1.2.14 and 1.4 first release there was functional but now the phone operator response number unavailable.

I have try to compose number between Cisco 7941 and Grandstream GXP 2000 with rfc2833, inband, auto and setting various mode (via sip, audio, rfc inside the phone)

What difference there is between two syntax?

exten => _0XXXXXXXXXX,1,Dial(SIP/203/${EXTEN:1}) ;with /
exten => _0XXXXXXXXXX,n,Hangup

exten => _0XXXXXXXXXX,1,Dial(SIP/203,3,D(${EXTEN:1})) ; with ,n,D()
exten => _0XXXXXXXXXX,n,Hangup


assuming you are using fxo ports on the ata-

the first one (sip/203/exten) sends the number in the sip packet. It says, ‘hello we have a call for you 203, and we want you to send it to (exten)’. The ata says ‘okay hang on a sec’, picks up the line, dials the number, and then pipes whatever the result is thru to *.

the second one just asks for the line. it says ‘hello we have a call for you 203’, at which point the ATA connects and pipes through the dialtone. Asterisk then dials (sends DTMF for) the exten which presumably dials the line.

generally dtmfmode= should be set to rfc2833. also update the firmware on your GXP2000 and ata. you can get FW at .

if that doesnt help, post what comes up on the CLI when you try to make it work…

Yes, the SIP/203 is the FXO port :exclamation:
Ok, i have solved with this one:

exten => _XXXXXXXXXX,n,Dial(SIP/203,60,D(w${EXTEN}))

because this:

exten => _XXXXXXXXXX,n,Dial(SIP/203/w${EXTEN})

don’t work, the second example play number directly without pause before.

Thankyou for the explain, it’s very clear. Probably the second example function well on digium card…

Anoher question, is it possible change the number of the phone line in exit to PSTN?

I have try with this and the number effectively change, but the line number rest the same

exten => _XXXXXXXXXX,1,Set(CALLERID(num)=0423123456)
exten => _XXXXXXXXXX,n,Set(CALLERID(rdnis)=0423123456)
exten => _XXXXXXXXXX,n,Set(CALLERID(all)=Manuel <0423123456>)
exten => _XXXXXXXXXX,n,NoOp(calleridnum : ${CALLERID(num)})
exten => _XXXXXXXXXX,n,NoOp(calleridrndis : ${CALLERID(rdnis)})
exten => _XXXXXXXXXX,n,NoOp(calleridall : ${CALLERID(all)})
and now the external call :wink:
exten => _XXXXXXXXXX,n,Dial(SIP/203,60,D(w${EXTEN}))

Anyway thankyou, and good week end :wink:

no you cannot set your outbound caller id on an analog line, this is because th eonly signalling that goes across an analog channel is DTMF, and caller id is set by the switch on the other side.

You could look in your ATA and turn on dialtone detection, or tweak its dialplan to pause before dialing, that would let the second (more correct) config example work…

Yes i have already look at the ata tone detection, but when change the parameter of the tone for the italian PSTN doesen’t work in any mode :imp: :imp: :imp:

At the end the HT 488 don’t take hang up (for some internal) and busy lines (sometimes), but for the price…

Thankyou for the reply!