[Freeswitch-users] Calls ending after around 5min
Oz Mortimer
oz at o-tec.tv
Wed Feb 15 11:39:25 MSK 2012
Hi Fraser,
Your console output in pastbin isn't complete so of limited use. I'd really
need to see you dialplan xml along with possibly your js - feel free to pm
me. You dialplan seems overly complicated too, you should be able to all of
that within a single js application.
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fraser
Redmond
Sent: 14 February 2012 21:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Calls ending after around 5min
I thought that too, and had done a text-search my code for sched_hangup, and
there's no mention of it anywhere.
Is there anything else that would set it based on other variables being set?
The closest things I have is I'm setting api_hangup_hook to run some
javascript when the call is hung up, and hangup_after_bridge=false
My dialplan is fairly complicated (several js files get run to do database
logging, business logic, etc.) So the console output is probably more
helpful:
http://pastebin.freeswitch.org/18385
Cheers,
Fraser
On 14 February 2012 15:41, Michael Collins <msc at freeswitch.org> wrote:
Like omortimer says it looks suspicious. I'd pastebin your dialplan and the
console log of a single call coming in (or going out) and let's see if
you've got a sched_hangup in there somewhere.
-MC
On Tue, Feb 14, 2012 at 12:26 PM, Fraser Redmond <fraserredmond at gmail.com>
wrote:
I've done a test call, and got a pcap of it, along with a sip debug. (I can
email you files if you are willing to take a look.)
There's no sip traffic until it gets to the log lines I pasted before, which
is followed by the server sending a BYE to the gateway and client.
I did a google search on "nat timeout", but the only thing relevant seemed
to be the rtp-timeout and session-timeout which I mentioned before.
I really think there's a setting somewhere telling it to hangup. The logs
have things like this as the call is being hung up (but before the BYE's are
sent) which seem suspicious
> Added task 4 switch_ivr_schedule_hangup (uuid) to run at 1329249352
> ending bridge by request from read function
Cheers,
Fraser
On 14 February 2012 14:08, Michael Collins <msc at freeswitch.org> wrote:
I would get pcaps of the calls that drop at 5 minutes, including media. If
you need suggestions on how to collect pcaps then check out this page:
http://wiki.freeswitch.org/wiki/Packet_Capture
Lots of tools at your disposal! Don't forget that Wireshark has a really
nice interface for exploring SIP calls and RTP streams.
-MC
On Tue, Feb 14, 2012 at 10:16 AM, Fraser Redmond <fraserredmond at gmail.com>
wrote:
I've got calls that are getting disconnected after 5 minutes (.
There's some calls that don't, and we don't have many calls that last that
long (which is why we hadn't noticed until now.)
I've had a look through the documentation, and the only thing I saw that
looked likely was the rtp-timeouts which I changed to this without any
change:
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
(changed in internal and external profiles)
Here's the log output for when the call ends:
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.340991 [DEBUG]
switch_core_session.c:1009 Send signal
sofia/internal/sip:12605 at 99.237.220.56:3907 [BREAK]
2012-02-14 17:42:58.340991 [DEBUG] switch_scheduler.c:214 Added task 4
switch_ivr_schedule_hangup (9d39d866-5732-11e1-b1a8-05fc5f64fa97) to run at
1329241378
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.361011 [DEBUG]
switch_core_session.c:726 Send signal
sofia/internal/sip:12605 at 99.237.220.56:3907 [BREAK]
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG]
switch_ivr.c:589 sofia/internal/sip:12605 at 99.237.220.56:3907 Command Execute
respond(603)
9d39d866-5732-11e1-b1a8-05fc5f64fa97 EXECUTE
sofia/internal/sip:12605 at 99.237.220.56:3907 respond(603)
9d39d866-5732-11e1-b1a8-05fc5f64fa97 2012-02-14 17:42:58.521032 [DEBUG]
mod_sofia.c:2363 Responding with 603 [Decline]
Any other settings I should be looking at?
Cheers,
Fraser
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