|Summary:||ASTERISK-20174: Asterisk becomes unresponsive when tryung to send fax with T38|
|Reporter:||Andrew Nowrot (andrutto)||Labels:|
|Date Opened:||2012-07-26 10:05:22||Date Closed:||2012-08-29 08:59:46|
|Environment:||Debian wheezy witj kernel 3.1.6 no PSTN cards||Attachments:|
|Description:||When trying to send fax with T38 between two fax devices connected to SIP ATA (SPA2102) my asterisk becomes unresponsive. ATA has a parameter "FAX Tone Detect Mode" it comes with three choices "caller and callee", "caller only", "callee only". When it is set to something other than "callee only" and I try to send a fax, asterisk freezes. It stops responding to anything, it is not processing calls and CLI is completely unresponsive. With "calee only" everythig works like charm. Is this a bug?|
I am sending faxes from a regular fax device to a machine which gives me the choice between fax transmission and leaving the voice message, so my side (calling) has to initiate the fax transmission. According to https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway only callee should initiate the T38 transmission and that is the properly configured endpoint. But when caller initiates the fax transmission Asterisk should not crash.
When doing sip debug this is the last line and after that asterisk freezes
Got T.38 offer in SDP in dialog firstname.lastname@example.org
Capabilities: us - (alaw|slin), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
|Comments:||By: Rusty Newton (rnewton) 2012-07-26 17:58:21|
Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).
This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!
Be sure to include the full log with DEBUG and VERBOSE of level 5, plus Additionally, include a packet trace of the SIP,RTP,UDPTL traffic during the issue occuring.
By: Matt Jordan (mjordan) 2012-08-29 08:59:41.730-0500
Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines