Larry Linde
2005-Jan-18 11:36 UTC
[Asterisk-Users] DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP
I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a zap channnel via a CAC ABII -> T100p (zaptel driver from same date as asterisk.) The call goes through just fine. Lets say I call a PSTN phone. I answer the PSTN I can talk to/from the zap channel like normal. If I want to send DTMF from the ZAP to the PSTN asterisk mutes the DTMF tones even after connect. I am running G711-ulaw for a codec to/from the PSTN(SIP) calls both incoming/outgoing to/from asterisk SIP or ZAP work fine for voice. But DTMF will not passthru. If I bypass asterisk and go direct to the MAXTNT from something like kphone. It works fine. DTMF is great. If I use kphone via asterisk -> PSTN it gets muted. If I call in from the PSTN to a ZAP channel or a SIP channel via asterisk it muted the DTMF tones. I have tried to switch dtmfmode for the SIP channels to inband/rfc2833/info No difference other then it can break the voicemail decode under asterisk in info. If I switch asterisk->MAXTNT to RFC2833 I can get DTMF to pass on key RELEASE for a short tone. It does not generate a tone on key press or hold. (Which is what I need {a long DTMF tone for a door release}) I must be missing something here, Asterisk can't be that broke that it won't pass DTMF correctly. BTW. I think it works fine if i gate out of asterisk via a T100P (PRI) and not a SIP channel. But I know the SIP gateway is working fine because direct to/from the TNT (kphone,ser,cisco,polycom) works. So its something is asterisk. How do I tell it to get the heck out of the way and let my data be free :) {sip.conf} [maxtnt-gw] nat=no qualify=yes type=friend canreinvite=no ; Asterisk by default tries to redirect the host=x.x.x.x context=maxtnt reinvite=no canreinvite=no disallow=all allow=ulaw dtmfmode=inband ; Choices are inband, rfc2833, or info accountcode=maxtnt {zapata.conf} context=analog group = 2 signalling=fxo_ks channel => 1-24 context=testing callerid="test #"<(612) 111-2222> channel => 1 {zaptel.conf} span=1,1,0,esf,b8zs fxoks=1-24