Well, I can say that possibly your vars.xml file got reset to default.
Can you check that file for clues?
-MC
Sent from my iPhone
On Nov 6, 2008, at 11:04 PM, Baskar <[EMAIL PROTECTED]> wrote:
Hi Michael,
Actually the recording file will be in the path /usr/local/
freeswitch/recordin
Hi Michael,*
Actually the recording file will be in the path
/usr/local/freeswitch/recordings.
I get those recoding file till 4th Nov 2008 i get the sound file in the
format 1001-00-2008-11-04-15-17-54.
On 5th Nov 2008 onwards there is no sound file created.
An
What is actually happening? Is there no sound file created or is it
something else wrong?
-MC
Sent from my iPhone
On Nov 6, 2008, at 9:59 PM, Baskar <[EMAIL PROTECTED]> wrote:
Hi,
Recoding is done through default.xml.
For past 1 month recoding is working fine.
Suddenly for p
Hi,
*
Recoding is done through default.xml.
For past 1 month recoding is working fine.
Suddenly for past 3day recording is not working.
I did not modify any thing in deafult .xml .
*
* My deafult.xml file*
OK, Brian,
It's been sent to your freeswitch email account.
The warnings are in the attached txt file. The msvc IDE data for the error
crashing FS is in the body of the email.
Please, let me know if you need more.
Mark
-Original Message-
From: Brian West <[EMAIL PROTECTED]>
To:
gmail got caught in the cross fire of the mail list server move
lastnight... they dont' like it when you change IP's apparently. It
should be flowing fine as of this morning.
Can you forward me the warning?
/b
On Nov 6, 2008, at 4:22 PM, [EMAIL PROTECTED] wrote:
Anthony, look in your gmai
Anthony, look in your gmail account since I got a warning about my email being
to long to be placed here.
Mark.
-Original Message-
From: Anthony Minessale <[EMAIL PROTECTED]>
To: freeswitch-users@lists.freeswitch.org
Sent: Thu, 6 Nov 2008 6:22 am
Subject: Re: [Freeswitch-users]
Hello,
Couple of quick questions:
-- How do I keep FS from sending RPID? And strip the header? And
likewise from respecting it?
-- How do I add a PAI header -- example here of what I thought would
work -- ... but it doesn't seem
to work (proceeding action is to dial a resou
Brian West wrote:
> You turn on inbound late negotiation and make the decision in the
> dialplan.
>
But I'm trying to do this on outbound/originate sessions. So nothing's
going through the dialplan. :) I'm initiating the script with originate
commands, from the CLI or XML-RPC.
> We fixed th
On Nov 6, 2008, at 1:46 PM, Brian Wood wrote:
> Brian West wrote:
>>
> But what if I want to force a specific codec?
You turn on inbound late negotiation and make the decision in the
dialplan.
>>> This script should beep, record until you stop speaking, beep again,
>>> pause, beep, playback,
Brian West wrote:
> When you're doing a playback with the native codecs you shouldn't put
> any extension on it. It will automagically select the file based on
> the current codec of the channel.
>
> filename = "/tmp/recording"
>
> Then have a file with all the codecs you support then FreeSWIT
This is svn trunk? There is no reason this should not work. it happens all
the time where this setting breaks it for people going the other way when
they don't want it to happen.
If you can't get it working we can probably configure it for you.
On Thu, Nov 6, 2008 at 11:55 AM, David Aldworth
When you're doing a playback with the native codecs you shouldn't put
any extension on it. It will automagically select the file based on
the current codec of the channel.
filename = "/tmp/recording"
Then have a file with all the codecs you support then FreeSWITCH will
pick the right one.
I'm doing some more testing with various codecs. I'm seeing some more
weirdness with the selection of a codec by file extension. I'm testing
this on FreeSWITCH trunk 10270.
The following script works with the .wav and .gsm extension, but
everything else seems to fail.
Using lowercase names (su
No love. They set extern ip so the IP comes through correctly, but the
acl did not seem to have any affect. We are still sending to the wrong
port. Sip trace, acl.conf.xml and sip profile are below:
U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060
SIP/2.0 100 Trying.
Via: SI
you are missing execute-app-arg
sendmsg
call-command: execute
execute-app-name: intercept
execute-app-arg:
On Thu, Nov 6, 2008 at 11:18 AM, Dennis <[EMAIL PROTECTED]> wrote:
> but what could be the cause, that there is no uuid in the intercept?
> we are sending the uuid of the outbound with t
but what could be the cause, that there is no uuid in the intercept?
we are sending the uuid of the outbound with the intercept.
as you can see under http://pastebin.freeswitch.org/6019 in the "test
intercept" part, we send the message with the uuid and get the answer
below (with intercept, outbo
Viktor,
For the user channel its user/[EMAIL PROTECTED] ... as for the rest
can you show me your entire config?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
/b
On Nov 6, 2008, at 9:45 AM, x y wrote:
Hi!
I'm new to freeswitch, and I'm trying to make an att_xfer in a
di
in your intercept example there is no uuid (note the empty () where the
uuid should be and the syntax error)
1. 2008-11-06 16:17:26 [DEBUG] switch_ivr.c:391
switch_ivr_parse_event()sofia/internal/
[EMAIL PROTECTED] Command Execute intercept()
2. 2008-11-06 16:17:26 [ERR] mod_dptools.c:
hi anthony,
i just pasted the results of 3 unsuccessful tests into the pastebin:
http://pastebin.freeswitch.org/6018
perhaps you could be so nice and have a look at it...
in the first test i made an uuid-bridge over our socket script.
in the second test i made an originate and then an intercept
Hi!
I'm new to freeswitch, and I'm trying to make an att_xfer in a dialplan, but
instead of giving a sofia/${domain}/${called_number} as , i
would like to use a loopback/${called_number}, because i would like to transfer
the call not just to different extensions. Is the
doh,
I keep doing that sorry.
apply-nat-acl not apply_nat_acl
On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <[EMAIL PROTECTED]> wrote:
> Yes. Below are settings that have been persistent through recent testing.
> Is there anything else we can try or should we open a jira?
>
>
>
>
proxy-media mode is still in the media path.
bypass-media is the one where you are not in the media path.
On Thu, Nov 6, 2008 at 9:30 AM, shehzad p <[EMAIL PROTECTED]> wrote:
>
> What if I need to stay in media path in some cases.
> Is there a way to prevent this issue?
>
>
> Anthony Minessale-2
What if I need to stay in media path in some cases.
Is there a way to prevent this issue?
Anthony Minessale-2 wrote:
>
> You could use proxy_mode to avoid messing with the SDP at all.
>
> set the option inbound-proxy-media to true in the profile and FS will not
> get involved in the codecs
>
On Nov 6, 2008, at 10:08 AM, Klaus Teller wrote:
> OK. I updated and tried flushing the DTMFs before playing the
> commands and it works. Thanks.
>
> Now, i feel there is a more general issue of scalability around DTMF
> (both inband as well as RFC2833) handling in Freeswitch. What do you
>
OK. I updated and tried flushing the DTMFs before playing the commands and it
works. Thanks.
Now, i feel there is a more general issue of scalability around DTMF (both
inband as well as RFC2833) handling in Freeswitch. What do you guys think?
What i've been working on is a tool for testing voic
Hi Brian, Anthony
Thank you for the help.
Freeswitch is GREAT!
Thanks,
Woody
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Was this not helpful?
http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk
/b
On Nov 6, 2008, at 8:23 AM, Peter P GMX wrote:
> It's rather simple
> - Setup a sip user on asterisk with username/password
> - Setup a gateway in freeswitch with the asterisk user credentials
> (ip,
>
It's rather simple
- Setup a sip user on asterisk with username/password
- Setup a gateway in freeswitch with the asterisk user credentials (ip,
username, password of asterisk)
- Define a route in the dialplan (e.g. default.xml) to route certain
numbers to the asterisk gateway
e.g.
You shou
Yes. Below are settings that have been persistent through recent
testing. Is there anything else we can try or should we open a jira?
On Nov 6, 2008, at 7:01
can you please download and build the development snapshot instead of the
1.0.1 since we are almost to 1.0.2 now.
http://files.freeswitch.org/freeswitch-snapshot.tar.gz
On Wed, Nov 5, 2008 at 10:05 PM, <[EMAIL PROTECTED]> wrote:
> Yup, I'm using ps_pizza.js.
>
> Mark
>
> -Original Messag
You could use proxy_mode to avoid messing with the SDP at all.
set the option inbound-proxy-media to true in the profile and FS will not
get involved in the codecs
On Thu, Nov 6, 2008 at 3:57 AM, shehzad p <[EMAIL PROTECTED]> wrote:
>
> In the following trace,
> aaa.aaa.aaa.aaa = originator
> bb
Never forget we have equal rights for call legs here. =D Things you do on
the inbound leg do not always carry to the outbound leg.
each leg is it's own channel and the variables do not carry across call legs
unless you use "export" like Brian said or
put the variable def inside {} of the dial strin
If that doesn't work one of your uuids are wrong or contains a superfluous
space or some other character
You have to understand that the inbound call that uses "socket" is the same
thing as &park()
They are both in a park state at that point and there is no functional
difference.
once both channel
did you remember to add
to the profile in question and restart?
On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <[EMAIL PROTECTED]>wrote:
> Brian, we updated the acl to:
>
>
>
>
>
> And the ACK is still going to the wrong (right but wrong) ip/port.
>
> Is there any way to get t
Try export instead of set here.
/b
On Nov 6, 2008, at 7:30 AM, Woody Dickson wrote:
>
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Hi,
I tried to avoid Freeswitch from playing moh when the meta_app is executed
by setting hold_music=silence:
However, when #1 is pressed, Freeswitch still attempts to play moh:
92.168.1.101 Processing meta digit '1' [execute_extension::record XML
features]
2008-11-07 05:18:30
Hi Dennis,
No, sorry, my knowledge of FreeSWITCH is still rather limited.
Hopefully someone else on the list will know.
Cheers, Birgit
On 06/11/08 09:14, Dennis wrote:
> hi birgit,
>
> thanks for your reply!
>
> some days ago anthony told me about "intercept" in the irc, but i seem
> not to b
the latest build brings the following errors:
2008-11-06 08:20:42 [CRIT] switch_loadable_module.c:767
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_dingaling.so
**/usr/local/freeswitch/mod/mod_dingaling.so: undefined symbol:
switch_core_hash_init**
2008-11-
In the following trace,
aaa.aaa.aaa.aaa = originator
bbb.bbb.bbb.bbb = freeswitch
ccc.ccc.ccc.ccc = terminator
#
U 2008/11/06 03:26:22.208150 aaa.aaa.aaa.aaa:5060 -> bbb.bbb.bbb.bbb:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0.
Vi
You need Mono 2.0 or later. My guess is that these particular errors are caused
by not using a C# 3.0 compiler.
-Michael
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Helmut Kuper
Sent: Thursday, November 06, 2008 2:05 AM
To: freeswitch-users@lists.fre
hi birgit,
thanks for your reply!
some days ago anthony told me about "intercept" in the irc, but i seem
not to be able to use it, at least not for connecting two channels
with each other.
we need to first originate, because we want to have complete control
over all channels and chose, what to d
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Michael,
I tried to compile mod_managed on centos 5.2/32bit, mono 1.2.4 and
mono-devel files. I get this error:
making all mod_managed
Loader.cs(63,58): error CS1026: Expecting ')'
Loader.cs(63,67): error CS1002: Expecting `;'
Loader.cs(69,14): e
I have done a little bit of maintenance on our list server tonight.
Things will level out as DNS starts to settle down.
Please report any issues to me directly off list please.
/b
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