you neet to get pcaps of the calls and look at the rtp and sip going on. On Wed, Jun 30, 2010 at 9:39 AM, Dan <freeswitch-users@digitaldan.com>wrote:
Thanks for your response, I put everything up on pastebin http://pastebin.freeswitch.org/13322 . The application in question is actually javascript, I'm using lua in production but was switching to the posted js version with the upgrade.
Now that I posted it i realized I have
<action application="set" data="rtp_timer_name=none"/>
in my dial plan, I believed I used it in the older version to get around some dtmf issues or choppy playback (can't remember), not sure if this could be part of the issue (although it works fine in the production version I'm running)
So I pulled one of the recordings that hung up after 4 minutes, but was only 24 seconds long, it sounded fine (but obviously too short). But another one that dropped after 5 minutes and only 19 seconds in length was very choppy and included short spurts of audio from parts of the call that were much longer then 19 seconds. ------------------------------ *From: *"Anthony Minessale" <anthony.minessale@gmail.com> *To: *freeswitch-users@lists.freeswitch.org *Sent: *Tuesday, June 29, 2010 12:54:21 PM *Subject: *Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT
it's not 100% accurate in the media timeout. It would be too expensive to use actual timers, it uses the number of samples you should be getting from rtp and a number of loops where no media was received.
Migrating from svn 13000 range to GIT is a big step and you may have to adjust to some new behaviors. media_timeout may not even have existed that long ago I don't recall.
If you don't need media timeouts turn off the param or turn it up to longer.
On Tue, Jun 29, 2010 at 1:09 PM, Michael Collins <msc@freeswitch.org>wrote:
Pastebin your dialplan and the lua script for starters. Also, is it the 5300 that is responding with the media timeout? -MC
On Tue, Jun 29, 2010 at 10:15 AM, Dan <freeswitch-users@digitaldan.com>wrote:
Hi guys, I have been running two freeswitch boxes (13754M) that answer calls from a cisco 5300 (both on the same network) and records them to disk with a small lua application. This has been working well for the past few months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I reverted back to the old freeswitch version (on the same hardware) and have not been able to reproduce it in a test environment yet ( I continue to try). Below are the relevant lines from the log files for one of the calls:
2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/ nobody@192.168.21.4) Callstate Change ACTIVE -> HANGUP 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/nobody@192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal sofia/external/nobody@192.168.21.4 [KILL] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal sofia/external/nobody@192.168.21.4 [BREAK] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/nobody@192.168.21.4 Restore previous codec PCMU:0.
My configuration is bone stock, so the rtp timeout value is at 300, but I have some calls that have lasted only 4 minutes. One other piece of information is that on one of the recordings that was hung up after 4 minutes and 17 seconds the recorded file was only 24 seconds long (it stopped recording after the first 24 seconds) , so I'm assuming freeswitch did not think there were any rtp packets to record.
Any ideas on where to start debugging this? I have setup a new freeswitch box connected to the same 5300 to reproduce, but have not been able to generate the call volume ( there where around 30 calls being recorded) yet, but I'm working on it.
Thanks!
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-- Anthony Minessale II
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