I'm doing some outbound dialing, and want to use mod_vmd to detect if a live
person picks up or a voicemail picks up. I've read the wiki, and have been
playing around with the dialplan implementation and the lua implementation,
along with capturing the mod_vmdvmd::beep event.
Using the examples
2009/4/3 Michael Collins m...@freeswitch.org:
On Fri, Apr 3, 2009 at 7:11 AM, Brian West br...@freeswitch.org wrote:
Did it sound more like a machine gun?
/b
Comfort noise for General Douglas McArthur I guess...
I thought General Norman Scwarzkopf (Stormin' Norman) would have been
more
Matt,
No that¹s all mod_vmd does...
If you want to do a more advanced analysis of media stream coming from the
client mod_amd is available under a commercial license. This does media
analysis to determine machine vs humans based on a hand full of metrics that
are tunable. Contact me off list for
Ciao Giovanni,
I suggest to update the startskype.sh script by adding a su username
statement,
in this way:
instead of starting skype as
echo myskypeuser xxx | DISPLAY=:101 /usr/bin/skype --pipelogin
is better to do:
su unixusername -c echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype
Hi Brian,
Is NAT a known problem? Is there a work around? The messages on the lists
seem to imply other folks have this working ok behind NAT firewalls. What's
your recommendation for how I should proceed?
regards
Andy
-Original Message-
From:
Matthew Fong mattdf...@gmail.com wrote:
My question is, is there a way to use mod_vmd to detect if an answering
machine or human has picked up within the first 1-2 seconds after being
answered?
Probably not. If you have an algorithm in mind that would achieve this with a
high degree of
Hi,
I've been using the record_session feature and wish to use PCMU or ul audio
format, but when I try to play back the audio in either format, it sounds
high-pitch and fast as if it is playing back at 2x speed. I looked at the
waveform recorded in PCMU and ul versus what it looks like when I
Preferably GUI based.
Thanks,
David
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I have been trying to setup 2 DID's to route to 2 extensions but
whenever I try it, the second configured DID always routes to the first
extension.
In my public.xml I have the following:
include
context name=public
extension name=DID 1
condition field=destination_number
On Tue, 2009-04-07 at 17:17 +1000, Jason White wrote:
Matthew Fong mattdf...@gmail.com wrote:
My question is, is there a way to use mod_vmd to detect if an answering
machine or human has picked up within the first 1-2 seconds after being
answered?
Probably not. If you have an algorithm
Hi,
I've been using the record_session feature and wish to use PCMU or ul audio
format, but when I try to play back the audio in either format, it sounds
high-pitch and fast as if it is playing back at 2x speed. I looked at the
waveform recorded in PCMU and ul versus what it looks like when I
Hi,
I've been using the record_session feature and wish to use PCMU or ul audio
format, but when I try to play back the audio in either format, it sounds
high-pitch and fast as if it is playing back at 2x speed. I looked at the
waveform recorded in PCMU and ul versus what it looks like when I
svn commit -mskypiax: modified configs/startskype.sh to specify which
unix user will start the Skype client instance. Thx to
mbrancale...@voismart.it
Sendingconfigs/startskype.sh
Transmitting file data .
Committed revision 12937.
:-)
On Tue, Apr 7, 2009 at 10:13 AM, Matteo
I want to do the following:
Due to missing Softclient with TLS/SRTP support on my Linux laptop
(Zoiper is almost there, but not yet with SRTP) I want to install a
local FS to listen on a local IP and then communicate via TLS/SRTP to my
FS in the Office.
As My Laptop has changing IPs (e.g.
On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote:
1st Question: Is that possible or is another solution preferrable?
Just use FreeSWITCH with mod_portaudio.
2nd Question: How can I change the amount of memory FS tries to
reserve
to an absolute minumum (I only have 1 call at a time).
Hi Guys,
I'm no Linux guru, but today I inadvertently had 1000+ call attempts
going through FS, load according to TOP was 16.5. Calls were still
absolutely perfect. Can I throw out the rule book on load ? CPU was
~45% on each core. (dual)
Regards,
Thanks Brian,
what I was actually looking for was to use a standard SIP soft phone
with some additional features.
I finally manged to make FS listen on 127.0.0.1 the following way:
vars.xml
X-PRE-PROCESS cmd=set data=domain=127.0.0.1/
internal.xml
param name=rtp-ip value=127.0.0.1/
I want to use a low bandwidth codec. But whenever I try to use speex I
get an error in the conference. We have FS trunk 1288. Switching back to
PCMx it works again.
Is there any problem with speex and DTMF or with transcoding?
Best regards
Peter
2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2624
2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
Opening File
[/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav]
[System error : No such file or directory.]
2009-04-08 00:43:24 [WARNING] mod_conference.c:4799
conference_function() Cannot ask the user
Chances are he just doesn't have the 16k sound files installed.
/b
On Apr 7, 2009, at 6:35 PM, Michael Collins wrote:
2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
Opening File
[/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav]
[System error : No such
Hi David,
Have seen a similar issue reported on whirlpool recently with another provider,
essentially if the ITSP does not forward the To: header with the correct
terminating DID you will not be able to determine the extention to route the
call to. Am I correct in saying you only have one
Hello,
I wrote an application using FreeSWITCH version 1.0.3, with mod_python and a
64 bit box on Red Hat.
The app works fine when one person dials in, but when a second person dials
in, the first call stops and waits until the second call is finished. It's
really strange - if the first call is
Our FreeSWITCH setup has an existing T1 using RBS to talk to a digital
modem pack in a Cisco 3845. I'm interested in changing from RBS to
ISDN. I changed both sides, restart things, and see FreeSWITCH report:
2009-04-07 18:53:15 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3
retries
I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a new
trunk and I have to go through vars.xml, etc changing $${local_ip_v4} like you
did.
Is there a way to change $${local_ip_v4} in one place. That way one wouldn't
have remember all the locations that it needs to be
mszla...@aol.com mszla...@aol.com wrote:
Is there a way to change $${local_ip_v4} in one place.
Of course. That's why it's a variable.
X-PREPROCESS cmd=set data=local_ip_v4=10.10.1.2/
this goes in vars.xml, substituting the desired address.
___
Hello,
I wrote an application using FreeSWITCH version 1.0.3, with mod_python and
a
64 bit box on Red Hat.
The app works fine when one person dials in, but when a second person dials
in, the first call stops and waits until the second call is finished. It's
really strange - if the first
Wonderful! Thank you sir.
-Original Message-
From: Jason White ja...@jasonjgw.net
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 7 Apr 2009 7:04 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1
and memory consumption
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