Summary:ASTERISK-09410: DTMF generation out of nowhere
Reporter:hristo (hristo)Labels:
Date Opened:2007-05-09 08:43:08Date Closed:2007-05-09 14:39:41
Versions:Frequency of
Environment:Attachments:( 0) rtp_debug.txt
Description:I have upgraded on Friday, last week to the latest 1.4 svn and I started getting this problem. On random calls Asterisk starts generating DTMFs all by itself and all that can be heard is a constant beep. There is no external message that may be triggering the DTMFs (either SIP signaling DTMF or other incoming RTP DTMF).

The problem appears randomly and for no obvious reason. Restart sometime helps for a while, but after 30 minutes or so almost all calls start getting DTMFs instead of sound. The DTMFs that I've noticed were "A" "B" "1" "2"...possibly others. None of the phones used in this tests has "A" "B" "C" "D" DTMF keypad, which is yet another proof that the DTMFs are generated by asterisk for some reason.

I have also tried the svn version from yesterday (mentioned in this TT) and the problem was still present. After a brief research I have noticed that there were two recent DTMF related changes to the 1.4 code (r62942 and r62789). I'm not sure if the problem was cased by them, but downgrade to SVN-branch-1.4-r62331 seems to solve the problem. Nothing special about this version - I simply had other server with this version, that seemed to work and downgraded all other servers to this known-to-work version.


Attached rtp debug, that clearly shows the DTMFs that are being sent. is X-Lite is Cisco gateway

For some strange reason it appears that the call is first put on hold, when 183 progress message is received (this is the in-band ringback tone coming from the Cisco):

[May  8 10:58:13] VERBOSE[3358] logger.c:     -- Call on SIP/domain-006e78b0 placed on hold
[May  8 10:58:13] VERBOSE[3358] logger.c:     -- Started music on hold, class 'default', on SIP/domain-00720fe0

But right after that, asterisk starts passing the progress:

[May  8 10:58:13] VERBOSE[3358] logger.c:     -- SIP/domain-006e78b0 is making progress passing it to SIP/domain-00720fe0

Not sure if this is related, but it doesn't appear in the logs from the know-to-work version.
Comments:By: Joshua C. Colp (jcolp) 2007-05-09 14:39:41

Fixed in 1.4 as of revision 63698 and trunk as of revision 63699.