Summary:ASTERISK-00201: DTMF not passed via chan_sip with early audio
Reporter:paul_cheng (paul_cheng)Labels:
Date Opened:2003-08-31 17:53:05Date Closed:2004-09-25 02:49:15
Versions:Frequency of
Description:This bug may be related to the other reports of * not responding to reload, restart now, etc. and artifact SIP channels. There are actually two bugs to report, so I will submit this report twice with each one separated by title.

When dialing a number which reaches an IVR system (usually toll-free numbers) via channel SIP and audio is received before the call is "answered", DTMF will not pass correctly. If the call is "answered" then DTMF passes correctly. If DTMFMODE=rfc2833 or DTMFMODE=info, then the IVR just keeps going and does not detect DTMF.

However, if DTMFMODE=inband and user presses a number, then the call goes blank and Asterisk does not hangup the call. Input at the console is then ignored for restart, reload, etc. however, sip show peers, etc. still work.

The only way to stop asterisk is to "kill" from a shell command line.


Using a SIP UA, dial 18008828880 (or 18004354000) via FWD or any other SIP Proxy which lets you connect an 800 call.

NOTE: Not all SIP UAs will accept early audio. Cisco ATAs will. Grandstreams will not.

Try DTMF using rfc2833 or info or inband.

If it is set to inband, asterisk will block and not release the channel.


Comments:By: John Todd (jtodd) 2003-10-16 04:13:49

Paul - is this still a problem?  I believe the other block issues were resolved, but please re-examine with a new CVS checkout to see if your particular bug still exists or not.

By: pcheng (pcheng) 2003-10-16 15:13:52

Will check when I have a free moment. A little swamped right now. Back soon.

By: pcheng (pcheng) 2003-10-16 15:18:30

Will check when I have a free moment. A little swamped right now. Back soon.

By: Brian West (bkw918) 2003-12-06 12:12:21.000-0600

I'm going to assume that this has been fixed ... If that isn't the case please find me on irc.freenode.net #asterisk bkw_