|Summary:||ASTERISK-13146: DTMF via INFO or RFC2833 via SIP for extensions processing in Asterisk 1.6 leads to instant channel hangup|
|Reporter:||Zam Loomstein (therealzam)||Labels:|
|Date Opened:||2008-12-01 01:58:12.000-0600||Date Closed:||2008-12-22 11:17:48.000-0600|
|Description:||Basically, calls are received via SIP from one of several providers (sellvoip, xo, voxbone). If DTMF is entered via SIP INFO or RFC2833, the channel instantly reverts to status 'UNKNOWN' and terminates.|
I've tried this on several different boxes with the same os and different hardware and the problem repeats.
A user entering the information to channel receives an almost instant hangup upon attempting to enter DTMF. With debugging on, the server aknowledges the dtmf as having been entered and then a SIP channel status change to 'UNKNOWN' followed by an instant hangup which appears more like a thread stop than anything else.
This problem does not occur on the latest released 1.4 version (not including svn releases, which wern't checked here).
Of potential consequence is that the system uses realtime/mysql for extension storage.
I'm not sure what logs are needed for debugging of this dtmf issue or how to get them. I can easily get his information though.
|Comments:||By: Joshua C. Colp (jcolp) 2008-12-08 10:17:03.000-0600|
Please provide complete console output with rtp debug, along with what DTMF you entered and what you were expecting to see. I suspect this is some sort of configuration issue.
By: Leif Madsen (lmadsen) 2008-12-22 11:17:48.000-0600
We require input from the original reporter into move issues forward. As it has been 2 weeks since a request for debugging information has elapsed, I'm closing this issue. If you have obtained the required information, please feel free to find a bug marshall in #asterisk-bugs on the Freenode IRC network at irc.freenode.net. Thanks!