[Asterisk-Users] ISDN debugging and SIP dial-in issue]

Mon Nov 17 06:08:30 MST 2003

Try this - change 
exten => s,3,Background(transfer)    ; schaefer

to
exten => s,3,Playback(transfer)    ; schaefer

and then dial 3 from your GS.

You should also add to sip.conf for [17476691152]:
disallow=all
allow=ulaw
allow=alaw

Are you sure you need the dtmfmode=inband for the GS? I don't have a GS, 
so look for GS samples on this list. In order to prevent codec problems 
and to allow transcoding you might also want to add canreinvite=no.

Also: Check your stripmsd= setting in modem.conf to make sure you are 
really dialing the mobile number you want to dial. Next to that I'd 
rather use the "new" syntax for the Dial application like 
"Dial(Modem/g1/012345,20,rt)".

Cheers, Philipp