[Freeswitch-users] Calls ending after around 5min

Wed Feb 15 20:28:11 MSK 2012

Hi Fraser,

Looking at the log you put in pastebin the call was good without the 5
minutes hangup. Can you put the complete log in pastebin where the call has
this problem.

The xml doesn't resemble the code snippet of the original 5 minute
disconnect. As I said before, I strongly suspect it is something in one of
the js files.

What file is calling the bridge to 12605 at .. As my guess is your issue is
here.

Cheers

Oz

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fraser
Redmond
Sent: 15 February 2012 16:16
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Calls ending after around 5min

So the background is that we run a call-center with a lot of detailed
tracking of the stages of calls for better monitoring of Rep's performance.
Later I hope to shift that to the event socket where it ought to be, but for
now, it's done inline in the dialplan using JS, often run asynchronously, so
as to not hold up the call while making a database insert.

We handle lots of different scenarios including inbound calls going to
queues, attended transfer of calls, and several other things. Some of these
are not implemented in the best manner due to the combination of 3 factors:

- inexperience with telephony (a lot of the code was done 2 years ago, with
bits upgraded since then. I had used asterisk for a couple of months before
that, and come from a webapp development background)

- lack of documentation for FS (most things are covered briefly, but many
things are only covered briefly)

- lack of time to work on implementing things properly (telephony is an
important part of our business, but other areas have needed development
more.)

The steps happening in our dialplan:

1) Website: database insert for the call, with status set to initialised

2) Website runs this command via ESL:

api originate
{callId=$callId,personId=$_PERSONID,agentExtn=$fromNum,language=en-us,Outbou
nd=Yes}user/$fromNum@$freeswitchDomain $toNumClean

(This dial's the rep's extn, then once they pick up it starts the bridge to
the external number.)

3) This hits the end of the attached dialplan (StaffeySwitchboard) which
runs a JS file

4) MainMenuCallsIn.js: sets some variables, and in this case sets
${destinationNumber} then transfers to "outbound_call XML outbound_call"
which then runs another JS file:

5) OutboundCall.js: does a manual version of LCR, by looking at the
phonenumber prefix and choosing a gateway and setting the variable
${dial_string}, then the dialplan transfers to "ConnectBLeg XML
DecideWhetherToCall"

6) That dialplan does several things:

   a) Runs BridgeBefore.js - in this scenario the only thing it really does
is set api_hangup_hook

   b) Runs api_Call_Stage_Update_SP.js in background (non-blocking so it
doesn't delay the call connection), which updates the database's call-status
to pickup delay (ringing / early-media )

   c) Exports some variables to the b-leg and start recording

   d) Sets api_Call_Stage_Update_SP.js to run when the call is connected to
update the database call-status to spoke-to (call connected / answered)

   e) Bridge to ${dial_string}

   f)  End recording and run BridgeEnded.js to set the database call-status 

   g) In other scenarios another action might be needed after a call
finishes, but in this case it doesn't do anything.

I've attached the JS files for steps 4 and 5, but not any of the ones in
(6), as they're all pretty small and simple.

Feel free to let me know anything you think I ought to change, I'm happy to
get any feedback you're willing to give. Though really I'm most interested
in solving this problem with calls getting hungup after 5 minutes.

Cheers,
Fraser

On 15 February 2012 03:39, Oz Mortimer <oz at o-tec.tv> wrote:

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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_________________________________________________________________________
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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_________________________________________________________________________
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consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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