The cause is a constrained stack space which can't be made larger if you aren't running FS as super user (you can reduce your stack limit, but once reduced can't increase it).  

I've also run into this in other situations.  I record calls as .wav files but then convert to mp3.  Initially, I tried to do the conversion via a system() api in the hangup hook.  But lame would die due to insufficient stack space.  Instead I now just move the .wav file to a processing directory where a cron job picks it up, extracts the metadata from the wav and then converts to mp3 applying the metadata.  

You could look at using an alternative sendmail.  Some smtp servers have a sendmail binary that is very lightweight and would not suffer in a low stack space environment.  postfix for instance just drops the email into the drop directory (which is then picked up by the postfix daemon).  

The real sendmail on the other hand is going to do everything in-process.  So it needs to do a lot more work to do that email delivery.

On Tue, May 11, 2010 at 12:56 AM, David Ponzone <david.ponzone@gmail.com> wrote:
Christian,

if you send the mail using a shell call to "sendmail", it's possible it won't work.
Some weeks ago, I and another person have noticed that when you do that, sendmail crashes.
I don't know what was the cause and if it is fixed.

Writing to console would be a simpler way to know your script is executed.

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




Le 11/05/2010 à 00:45, Christian Löschenkohl a écrit :

yes, and this is my problem
it looks normal but the scripts are not executed (i send a email in the fist line of
the script and there is no mail send to me here).
the call is hung up without much comment, the dialplan runs through until the hangup -
without the bridge command executed from the script.

attended transfer is here, with this it works
http://pastebin.freeswitch.org/12959

the scripts do also work if i call the extension directly.
the extension 50 also rings if i put the bridge in the dialplan - my only problem here
is that it looks like the script is called but it is never executed.

br



Anthony Minessale wrote:

it looks like it works fine to me.

the call transfers back to the dialplan where it again executes your
socket app and that app either does nothing
or hangs up.

See line 789 of your pastebin.


2010/5/10 Christian Löschenkohl <christian.loeschenkohl@xpirio.com
<mailto:christian.loeschenkohl@xpirio.com>>

   the scripts do some database lookups (setting callers name) and do
   find the called
   user e.g. expand the called 50 to 50@customer.domain.com
   <mailto:50@customer.domain.com> and finaly do the call bridge.

   the transfer itself is done by the phone/softclient with a refer
   packet, as far as i can see.

   a debug trace is here
   http://pastebin.freeswitch.org/12958

   br

   Anthony Minessale wrote:

what exactly does your script call to try to make it transfer?

2010/5/10 Christian Löschenkohl
   <christian.loeschenkohl@xpirio.com
   <mailto:christian.loeschenkohl@xpirio.com>
<mailto:christian.loeschenkohl@xpirio.com
   <mailto:christian.loeschenkohl@xpirio.com>>>

   hello

   i have a problem with a freeswitch hosted pbx system.
   when i call in to a extension (e.g. 40) an answer the call
   and then
   do a blind transfer to extension 50 it
   doesn't work.
   the outbound socket script do not work or do not get called
   in this
   case (no mail from the script).

   on the other hand if i do a attended transfer it works as it
   should
   (sending mail in the script works).
   the same context, the same scripts.

   we use snom 320 phones, the sip refer looks normal.
   blind transfer also doesn't work with a softclient (sjphone)

   where could i start?

   br

   --
   Ing. Christian Löschenkohl
   Technische Leitung, Forschung & Entwicklung VoIP

   xpirio
   Telekommunikation & Service GmbH
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   T  +43 (0) 5 77 11 - 1000
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   Technische Leitung, Forschung & Entwicklung VoIP

   xpirio
   Telekommunikation & Service GmbH
   Lakeside B04
   9020 Klagenfurt
   Austria

   T  +43 (0) 5 77 11 - 1000
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Anthony Minessale II

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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
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--
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl@xpirio.com

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