[Freeswitch-users] Calls ending after around 5min

Fraser Redmond fraserredmond at gmail.com
Wed Feb 15 19:16:03 MSK 2012 

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=$fromNum,language=en-us,Outbound=Yes}user/$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
>
> ****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120215/40709e29/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: default.xml
Type: text/xml
Size: 14638 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120215/40709e29/attachment-0001.xml 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: MainMenuCallsIn.js
Type: application/x-javascript
Size: 9074 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120215/40709e29/attachment-0002.js 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: OutboundCall.js
Type: application/x-javascript
Size: 7188 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120215/40709e29/attachment-0003.js

Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users mailing list