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). <div><br></div><div>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. </div>
<div><br></div><div>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). </div>
<div><br></div><div>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.<br><br><div class="gmail_quote">On Tue, May 11, 2010 at 12:56 AM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com">david.ponzone@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">Christian,<div><br></div><div>if you send the mail using a shell call to "sendmail", it's possible it won't work.</div>
<div>Some weeks ago, I and another person have noticed that when you do that, sendmail crashes.</div><div>I don't know what was the cause and if it is fixed.</div><div><br></div><div>Writing to console would be a simpler way to know your script is executed.</div>
<div><br></div><div><div class="im"><div> <span style="border-collapse:separate;color:rgb(0, 0, 0);font-family:Helvetica;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div style="word-wrap:break-word">
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<div><font face="'Helvetica Neue'"><font color="#1C00FF">David Ponzone </font><font color="#000000" size="3"><span style="font-size:12px">Direction Technique</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size:13px">email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></font></font></div>
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<span><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;text-align:justify;color:rgb(192, 192, 192)"><i>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. </i><b><i>IPeva</i></b><i> 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.</i></div>
<div style="text-decoration:underline;text-align:justify"><font color="#C0C0C0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br></div></span><br> </div><br></div><div><div>Le 11/05/2010 à 00:45, Christian Löschenkohl a écrit :</div>
<div><div></div><div class="h5"><br><blockquote type="cite"><div>yes, and this is my problem<br>it looks normal but the scripts are not executed (i send a email in the fist line of<br>the script and there is no mail send to me here).<br>
the call is hung up without much comment, the dialplan runs through until the hangup -<br>without the bridge command executed from the script.<br><br>attended transfer is here, with this it works<br><a href="http://pastebin.freeswitch.org/12959" target="_blank">http://pastebin.freeswitch.org/12959</a><br>
<br>the scripts do also work if i call the extension directly.<br>the extension 50 also rings if i put the bridge in the dialplan - my only problem here<br>is that it looks like the script is called but it is never executed.<br>
<br>br<br><br><br><br>Anthony Minessale wrote:<br><br><blockquote type="cite">it looks like it works fine to me.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">the call transfers back to the dialplan where it again executes your <br>
</blockquote><blockquote type="cite">socket app and that app either does nothing<br></blockquote><blockquote type="cite">or hangs up.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">See line 789 of your pastebin.<br>
</blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">2010/5/10 Christian Löschenkohl <<a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a> <br>
</blockquote><blockquote type="cite"><mailto:<a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a>>><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">
the scripts do some database lookups (setting callers name) and do<br></blockquote><blockquote type="cite"> find the called<br></blockquote><blockquote type="cite"> user e.g. expand the called 50 to <a href="mailto:50@customer.domain.com" target="_blank">50@customer.domain.com</a><br>
</blockquote><blockquote type="cite"> <mailto:<a href="mailto:50@customer.domain.com" target="_blank">50@customer.domain.com</a>> and finaly do the call bridge.<br></blockquote><blockquote type="cite"><br></blockquote>
<blockquote type="cite"> the transfer itself is done by the phone/softclient with a refer<br></blockquote><blockquote type="cite"> packet, as far as i can see.<br></blockquote><blockquote type="cite"><br></blockquote>
<blockquote type="cite"> a debug trace is here<br></blockquote><blockquote type="cite"> <a href="http://pastebin.freeswitch.org/12958" target="_blank">http://pastebin.freeswitch.org/12958</a><br></blockquote><blockquote type="cite">
<br></blockquote><blockquote type="cite"> br<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> Anthony Minessale wrote:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">
<blockquote type="cite">what exactly does your script call to try to make it transfer?<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">
2010/5/10 Christian Löschenkohl<br></blockquote></blockquote><blockquote type="cite"> <<a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a><br></blockquote><blockquote type="cite">
<mailto:<a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a>><br></blockquote><blockquote type="cite"><blockquote type="cite"><mailto:<a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a><br>
</blockquote></blockquote><blockquote type="cite"> <mailto:<a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a>>>><br></blockquote><blockquote type="cite">
<blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> hello<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite">
<blockquote type="cite"> i have a problem with a freeswitch hosted pbx system.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> when i call in to a extension (e.g. 40) an answer the call<br>
</blockquote></blockquote><blockquote type="cite"> and then<br></blockquote><blockquote type="cite"><blockquote type="cite"> do a blind transfer to extension 50 it<br></blockquote></blockquote><blockquote type="cite">
<blockquote type="cite"> doesn't work.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> the outbound socket script do not work or do not get called<br></blockquote></blockquote><blockquote type="cite">
in this<br></blockquote><blockquote type="cite"><blockquote type="cite"> case (no mail from the script).<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite">
<blockquote type="cite"> on the other hand if i do a attended transfer it works as it<br></blockquote></blockquote><blockquote type="cite"> should<br></blockquote><blockquote type="cite"><blockquote type="cite"> (sending mail in the script works).<br>
</blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> the same context, the same scripts.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote>
<blockquote type="cite"><blockquote type="cite"> we use snom 320 phones, the sip refer looks normal.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> blind transfer also doesn't work with a softclient (sjphone)<br>
</blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> where could i start?<br></blockquote></blockquote><blockquote type="cite">
<blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> br<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite">
<blockquote type="cite"> --<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Ing. Christian Löschenkohl<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Technische Leitung, Forschung & Entwicklung VoIP<br>
</blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> xpirio<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">
Telekommunikation & Service GmbH<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Lakeside B04<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> 9020 Klagenfurt<br>
</blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Austria<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">
T +43 (0) 5 77 11 - 1000<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> F +43 (0) 5 77 11 - 1002<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> E <a href="mailto:christian.loeschenkohl@xpirio.com" target="_blank">christian.loeschenkohl@xpirio.com</a><br>
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<br></blockquote><blockquote type="cite"> --<br></blockquote><blockquote type="cite"> Ing. Christian Löschenkohl<br></blockquote><blockquote type="cite"> Technische Leitung, Forschung & Entwicklung VoIP<br></blockquote>
<blockquote type="cite"><br></blockquote><blockquote type="cite"> xpirio<br></blockquote><blockquote type="cite"> Telekommunikation & Service GmbH<br></blockquote><blockquote type="cite"> Lakeside B04<br></blockquote>
<blockquote type="cite"> 9020 Klagenfurt<br></blockquote><blockquote type="cite"> Austria<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> T +43 (0) 5 77 11 - 1000<br></blockquote>
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AIM: anthm<br></blockquote><blockquote type="cite"><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a> <br></blockquote><blockquote type="cite"><mailto:<a href="mailto:MSN%253Aanthony_minessale@hotmail.com" target="_blank">MSN%3Aanthony_minessale@hotmail.com</a>><br>
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</blockquote><blockquote type="cite">IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> <<a href="http://irc.freenode.net" target="_blank">http://irc.freenode.net</a>> #freeswitch<br></blockquote>
<blockquote type="cite"><br></blockquote><blockquote type="cite">FreeSWITCH Developer Conference<br></blockquote><blockquote type="cite"><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> <br>
</blockquote><blockquote type="cite"><mailto:<a href="mailto:sip%253A888@conference.freeswitch.org" target="_blank">sip%3A888@conference.freeswitch.org</a>><br></blockquote><blockquote type="cite"><a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a> <br>
</blockquote><blockquote type="cite"><mailto:<a href="mailto:googletalk%253Aconf%252B888@conference.freeswitch.org" target="_blank">googletalk%3Aconf%2B888@conference.freeswitch.org</a>><br></blockquote><blockquote type="cite">
pstn:+19193869900<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">------------------------------------------------------------------------<br></blockquote>
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<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote><br><br>-- <br>Ing. Christian Löschenkohl<br>Technische Leitung, Forschung & Entwicklung VoIP<br><br>xpirio<br>Telekommunikation & Service GmbH<br>
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