[Home]

Summary:ASTERISK-16462: Some hanging ReceiveFAX app after a dax y with 1000 Fax
Reporter:Kristijan Vrban (vrban)Labels:
Date Opened:2010-07-29 11:06:54Date Closed:2011-06-07 14:05:10
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Resources/res_fax_spandsp
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I have an asterisk 1.8beta2 to receive fax with ReceiveFAX. After every working day after i received about 1000 fax, i have 1-3 hanging ReceiveFAX. They just do nothing, there is no UDPTL traffic any more. I can hangup them with the hangup cmd.

The questions is, how can i debug what happend to them? And perhaps we need also a "udptltimeout" option like the rtptimeout option to prevent this.

****** ADDITIONAL INFORMATION ******

spandsp is spandsp-20100725
Comments:By: Leif Madsen (lmadsen) 2010-07-29 11:21:41

You should use the asterisk-users mailing list for help with debugging this. To start you could probably capture information like DEBUG logs and perhaps output of things like lsof, etc...

By: Kristijan Vrban (vrban) 2010-07-29 11:24:25

wow, i just found the pcap's i record for this hanging ReceiveFAX, and the pcaps where still growing (240MB size!), because since hours there is outgoing a RTP stream from the asterisk 1.8, to my carrier.

examine more closely in the moment

By: Kristijan Vrban (vrban) 2010-07-29 11:27:10

hello, but there is also a bug in asterisk-1.8, whatever the original problem is.
The bug is, that "rtp set debug on", does not show the rtp which is going out from asterisk.



By: Kristijan Vrban (vrban) 2010-07-29 11:52:53

i was able to see, that this happen, after asterisk send out the T.38 re-INVITE, get a "100 Trying" (from the SIP proxy that i in between) but never get a answer then to the T.38 re-INVITE. And then asterisk/ReceiveFAX send an endless RTP stream out, an never hang up.

I try to help myself with rtptimeout and SIP timeout. And i dont now if this behave is a ReceiveFAX  bug. Or just a special situation, which rtptimeout and SIP timeout therefor are.

By: Leif Madsen (lmadsen) 2010-07-29 11:53:02

Please file the rtp set debug on issue as a separate issue.

By: Paul Belanger (pabelanger) 2010-09-04 14:16:39

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