Luis M. Zuccolo wrote:
> The problem is the "null" section
Yes, switch_simple_email is probably being called with a null first argument.
This shouldn't happen.
Which svn revision are you on? Does it still happen with the latest svn
revision?
___
Fr
Ok, my _javascript_, dialplan, etc has been Wiki'd...
The new page I created [Examples _javascript_ Conference IVR] is
here:
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
I linked it from the [_javascript_ Examples] page here:
http://wiki.freeswitch.org/wiki/_javascript__
Yes, in console works well (without null).
This variables are sets in switch.conf.xml:
Previously I've omitted the error:
sh: -c: line 0: syntax error near unexpected token `('
sh: -c: line 0: `/bin/cat /tmp/mail.124363022502bd | sendmail -t (null)'
The problem is the "null" section
Jason White wrote:
> See the mailer-ap and mailer-app-args variables in
> autoload_configs/switch.conf.xml and be sure they are set correctly for your
> installation. Try running the Postfix sendmail program manually to be sure
> that it is working correctly.
sendmail -t is the default, thus I
Luis M. Zuccolo wrote:
> I'm using postfix, that has a compatiblilty interface to sendmail.
I've used this with Sendmail successfully; it should work with Postfix too.
See the mailer-ap and mailer-app-args variables in
autoload_configs/switch.conf.xml and be sure they are set correctly for your
I'm using postfix, that has a compatiblilty interface to sendmail.
On Fri, 2009-05-29 at 21:36 -0500, Brian West wrote:
> Are you really using sendmail or are you using something like exim?
>
>
> /b
>
> On May 29, 2009, at 8:58 PM, Luis M. Zuccolo wrote:
>
> > Why the vm_mailto variable isnt't
If you happen to have a polycom or snom and you use the new sched_api
extension I added to trunk (commented out) it will sched_api and snag
the zrtp sas1 and sas2 strings and 4 seconds after the call is up
update the display of the polycom with those two strings... kinda
handy eh?
For thos
Brian West wrote:
> This is normal because the switch from clear to secure can happen
> quickly on one end or the other and you'll have a few packets that get
> thru before one end is ready... nothing to be worried about.
I thought that might be the scenario.
In a typical FreeSWITCH to FreeS
Are you really using sendmail or are you using something like exim?
/b
On May 29, 2009, at 8:58 PM, Luis M. Zuccolo wrote:
Why the vm_mailto variable isnt't passed to the script?
What's wrong?
Someone can assist me?
Thanks in advance
Luis Zuccolo
Brian West
br...@freeswitch.org
-- Meet us
Hi:
I get this error when voicemail try to send an email:
'/bin/cat /tmp/mail.12436473394319 | sendmail -t (null)'
This is the called extension:
Why the vm_mailto variable
I'll post the script, dialplan and how-to on the wiki as soon as I
can...
Will follow up on this thread with a link once it's complete.
Thanks for the interest.
Brian West wrote:
Maybe someone can do a wiki page with scripts and a howto?
/b
On May 29, 2009, at 3:31 PM, Anto
Maybe someone can do a wiki page with scripts and a howto?
/b
On May 29, 2009, at 3:31 PM, Anton Karpov wrote:
I'd be very interested to see your script and dialplan , for me it's a
very important issue as my conference server facing to outside and I
need to have moderators and regular users e
I'd be very interested to see your script and dialplan , for me it's a
very important issue as my conference server facing to outside and I
need to have moderators and regular users entering different pins.
Anton
jcro...@gmail.com wrote:
> Unfortunately, the instance of FreeSwitch where I've bee
On Thu, May 28, 2009 at 10:37 PM, Marc Orenberg wrote:
> Hi, I'd like FreeSWITCH to be able to communicate with a Musicam "Prima LT"
> device. (http://www.musicamusa.com/products/prima/PrimaLT.htm). This is a
> "POTS codec", which (I've just learned) means that the connection is made
> via regul
Can you try to do a binary search and nail down the exact version that
caused this issue and then file a bug on http://jira.freeswitch.org.
Thanks
Mike
On May 29, 2009, at 9:55 AM, Peter Olsson wrote:
I’m on Windows, so I have everything under my fs directory, but I
deleted the complete dir
Hi Anthony,
I updated to rev 13496 -- now I have a different problem... I connect to
the event socket interface, ask for all events... then never receive any
events!
>From telnet:
"
Content-Type: auth/request
auth ClueCon
Content-Type: command/reply
Reply-Text: +OK accepted
events plain all
Co
Hi Guys,
I have install WANPIPE Release: 3.5.2 for sangoma A104D and FreeSwitch
1.0.4pre8 with openzap modul.
and I use Lua script for playing wav file.
and I got error like this below while I call the number 0312982300, if i
run ./fs_cli , the FS can pickup a call for moment, after more than 1
Unfortunately, the instance of FreeSwitch where I've been playing with
this is at work and I can't get to it at the moment... Eventually, I
plan to post my entire implementation (_javascript_ and dialplan) on the
Wiki because I've added quite a few improvements to the confcall.js
example that
I'm on Windows, so I have everything under my fs directory, but I deleted the
complete directory and did everything from scratch...
/Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 29 maj 2009 1
Actually I deleted everything from disk and downloaded a fresh clean copy from
SVN and rebuilt it from scratch. I should mention that I'm on windows, so I
never do "make current". I just do a full clean, get latest from SVN and
rebuild, that's what I do every time. But for this time I even delet
did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too
?
On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <
peter.ols...@visionutveckling.se> wrote:
> Nope – it’s not :)
>
>
>
> Just to make sure I even deleted the source completely, and checked
> everything out again.
>
>
>
when you say "i did that"
you typed "make current" to rebuild?
or you are assuming your successful compile is the same effect as cleaning
the 100 object files
that have the wrong data structure in them so the audio data they really
seek is 8 bytes offset from where they think they are
until they a
Nope - it's not :)
Just to make sure I even deleted the source completely, and checked everything
out again.
/Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 29 maj 2009 15:26
Till: freeswitch-users@l
I did that, and it compiles fine. It's just not working :) But as I said in my
last post, I think it could also be related to sofia, when using h323 it
works... However - maybe I'm using opal's RTP stream by then..?
I'll get some logs for the scenario, and if I don't find a solution I'll start
Nope its not a sofia issue... its build skew ;)
/b
On May 29, 2009, at 8:24 AM, Peter Olsson wrote:
I've looked into this a bit more now, and I think it is a sofia
issue, I will look trough the changes in sofia since I had the last
working configuration, and see if I find anything.
/Peter
I've looked into this a bit more now, and I think it is a sofia issue, I will
look trough the changes in sofia since I had the last working configuration,
and see if I find anything.
/Peter
-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch
I would recommend not turning on the SLA option then... I had to add
that in because when using TLS the phone would try to call the IP
which would fail because the SSL cert wouldn't match and the poor
phone would kill over, lock up and reboot sometimes :P GO POLYCOM!
With that option not
the upgrade changed the switch_frame structure so most likely you did not do
"make current" as we always recommend.
Try that..
On Fri, May 29, 2009 at 5:46 AM, Jason White wrote:
> Peter Olsson wrote:
> > After using the latest trunk revisions I get no audio anymore. The last
> > working b
Peter find me on IRC and let me into your machine so I can trouble
shoot this.
Thanks,
Brian
On May 29, 2009, at 7:58 AM, Peter P GMX wrote:
And mine with the same behaviour on Linux.
Best regards
Peter
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
This is normal because the switch from clear to secure can happen
quickly on one end or the other and you'll have a few packets that get
thru before one end is ready... nothing to be worried about.
/b
On May 29, 2009, at 5:02 AM, Jason White wrote:
After ZRTP negotiation is complete (the ZR
Its called build skew... we added an extra_data element to the frame
struct. Please do a fresh checkout and build.
/b
On May 29, 2009, at 5:46 AM, Jason White wrote:
Peter Olsson wrote:
After using the latest trunk revisions I get no audio anymore. The
last
working build I have is about
maybe you have a nat issue sending byes to the phones
enable debug log by pressing f8 or typing console loglevel debug
and turn on sofia trace with "sofia profile internal siptrace on"
capture the entire thing and paste it to
http://pastebin.freeswitch.org
On Fri, May 29, 2009 at 4:37 AM, Gopal
There is a very long explanation as to the differences behind asterisk AMI
and FreeSWITCH Event Socket that I will not get into now
but it's not related to APR or TCP socket performance whatsoever it's more
about asynchronous versus monolithic modeling.
On Fri, May 29, 2009 at 4:42 AM, Gopalakris
try enabling the Path header too
we fully support that
On Fri, May 29, 2009 at 1:37 AM, Jim Burke wrote:
> Thanks Brian, will check it out.
>
> I am using FS as Voicemail behind Opensips. As we have 2 Opensips
> servers if FS responds with a Contact header with a URI value we
> cannot route th
And mine with the same behaviour on Linux.
Best regards
Peter
Diego Toro schrieb:
> Hi, my job with FS has been on Windows.
>
> Diego
>
> --- On *Thu, 5/28/09, Brian West //* wrote:
>
>
> From: Brian West
> Subject: Re: [Freeswitch-users] The calls are dropped during register
> To:
Hi, my job with FS has been on Windows.
Diego
--- On Thu, 5/28/09, Brian West wrote:
From: Brian West
Subject: Re: [Freeswitch-users] The calls are dropped during register
To: freeswitch-users@lists.freeswitch.org
Date: Thursday, May 28, 2009, 8:22 PM
Anything on linux?
/b
On May 28, 20
Peter Olsson wrote:
> After using the latest trunk revisions I get no audio anymore. The last
> working build I have is about 5 days ago. I havn't upgraded until today, so
> I don't know exactly when this happened.
You could always check out some intermediate revisions, compile them, and see
if t
Sorry for missing this in my last post, but I'm using sofia for all calls.
/Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Peter Olsson
Skickat: den 29 maj 2009 12:31
Till: 'freeswitch-users@lists.freeswitch.org'
Ämne: [Frees
After using the latest trunk revisions I get no audio anymore. The last working
build I have is about 5 days ago. I havn't upgraded until today, so I don't
know exactly when this happened.
I've noticed quite a few changes on the RTP stack, beacuse of the
implementation om ZRTP, and I guess it's
After ZRTP negotiation is complete (the ZRTP state machine has entered the
"secure" state), I get a number of lines in the log as follows (FreeSWITCH
rev. 13501):
2009-05-29 16:43:19 [DEBUG] switch_rtp.c:538 zrtp_logger() [zrtp protoco]:
ERROR! Decrypt failed. ID=14:DH s=SRTP authentication failu
Ok Thanks Mike.
I hope in asterisk it is not there. I was trying some couple of things,
1. when I use Java servlet to dial a call thru asterisk using manager
interface it slows down.
2. when I use the same Java servlet with Freeswitch I feel the speed and
usage in the tomcat. Its bit faster for s
Sorry for my presentation.
Call - is nothing but the outbound number in the far end. (this is the leg I
am trying to transfer)
Extension - is nothing but the internal softphones
My internal extension are not hanging up.
On Thu, May 28, 2009 at 8:51 PM, Anthony Minessale <
anthony.miness...@gmail.c
I could not get this working on current trunk. Can you post your
configuration on conference module and the dialplan example?
Thanks,
jmesquita
On Thu, May 28, 2009 at 12:56 PM, Michael Collins wrote:
>
>
> On Wed, May 27, 2009 at 8:00 PM, j3flight wrote:
>
>>
>> Wiki Tax paid...
>> That was m
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