[Freeswitch-users] blind transfer and outbound socket
Rupa Schomaker
rupa at rupa.com
Tue May 11 06:08:37 PDT 2010
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 at 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
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>
> 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 at xpirio.com
>
> <mailto:christian.loeschenkohl at 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 at customer.domain.com
>
> <mailto:50 at 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 at xpirio.com
>
> <mailto:christian.loeschenkohl at xpirio.com>
>
> <mailto:christian.loeschenkohl at xpirio.com
>
> <mailto:christian.loeschenkohl at 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
>
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>
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>
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>
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> E christian.loeschenkohl at xpirio.com
>
> <mailto:christian.loeschenkohl at xpirio.com>
>
> <mailto:christian.loeschenkohl at xpirio.com
>
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>
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>
> Anthony Minessale II
>
>
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> >>
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> >>
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> <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
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> >>
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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 at xpirio.com
>
> <mailto:christian.loeschenkohl at xpirio.com>
>
>
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> --
>
> Anthony Minessale II
>
>
> FreeSWITCH http://www.freeswitch.org/
>
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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 at xpirio.com
>
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--
-Rupa
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