I bit off topic but.
Using FS to send calls sip to the LD carrier.
Some calls have problems where they drop the call or audio drops or
whatever.
The carrier's first response is that we dropped the call. But this is
a day later after the trouble has been reported.
I am looking for guidance
Thanks. But when I made these entries in /etc/odbc.ini and rebooted.
[freeswitch]
Driver = MySQL
SERVER = 127.0.0.1
PORT= 4040
DATABASE= mydb
OPTIONS = 67108864
.I still get FS complaining with this.
Nov 27 08:45:57 P3 freeswitch[27933]:
Subject: Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
Are you using the myodbc 3.51.18 version or higher ?
I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to
upgrade from jaunty..
regards,
Leon
On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote:
Thanks. But when
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bind extention to a different Dialplan
andcdr php?
On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact fr...@impactfax.com
wrote:
Is there a way to bind a particular extension to a different dialplan
php and a different cdr php
Is there a way to bind a particular extension to a different dialplan
php and a different cdr php script than the default one?
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FreeSWITCH-users@lists.freeswitch.org
FS is in the media path of an IVR call.
At the moment, the call is ulaw with DTMF in the audio I think coming
into FS and leaving FS.
The call is coming from an Asterisk server and going to an Asterisk
server.
Is there a way to disable FS from passing DTMF at some point in the
call? For
We may have seen this also once before.
But in this case, it is 'needed' (or would be helpful) because the telco
is not doing the EC correctly.
I can see how the originating telco should be the one to fix the
problem. But it always seems that is easier said than done.
Is there no way at all
I have this version running on fedora 8 right now. Compiled fine and is
in production.
version
FreeSWITCH Version 1.0.trunk (10960)
However, I was in the src directory and ran make current and after
starting to compile it blew up with
Making all in libspeex
make[4]: Entering directory
: Thursday, March 26, 2009 10:27 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] compile poblem - FIXED_POINT
orFLOATING_POINT
make speex-reconf
On 26-Mar-09, at 10:19 AM, Frank @ Impact wrote:
I have this version running on fedora 8 right now. Compiled fine
I am trying to detect if a caller is an automated greeting voice. And
if so, take an action.
I have samples of the caller recording that I am looking to match.So
this is like a really complex tone detection I guess.
It would work like this.
- Call comes in
- We answer/bridge the call
-
Ok. Maybe it is more like answering machine detection in reverse?
Detection on the caller leg instead of the called leg.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent:
The two endpoints are sip (asterisk) and ulaw.
Thanks.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, December 29, 2008 2:56 PM
To:
Yes. I had tried that. Put a sleep 15 in the shell script before I
looked at the file. Same results however. FS just does not appear to
be closing that record file on hangup.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Mike,
I did some testing and this file is not getting closed. I called the
script on hangup. Made sure both legs hungup and then even did a sleep
for 5 secs to make sure FS could close any files is needed to. Then I
made a copy of the wav file to a tmp file. Then ended the script to
return
not a valid call from api_hangup_hook?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
On Dec 27, 2008, at 5:21 PM, Frank @ Impact wrote:
I was trying to stop a session record from
I was trying to stop a session record from lua but when I try I get a
Result is INVALID COMMAND!
I am calling this lua script with
action application=set data=api_hangup_hook=lua proc.lua ../
so by the time the lua is called, someone has hungup one of the legs.
In the lua script I am using
All I am passing into the script is the recording file name.
I tried using the system command right after the bridge command but
before a hangup command. Thusly,
action application=bridge data=${enum_auto_route}/
action application=system data=/root/procrecording.sh
...@lists.freeswitch.org] On Behalf Of
Frank @ Impact
action application=bridge data=${enum_auto_route}/
action application=system data=/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23__4_1.wav/
action application=hangup/
The problem I am seeing is that sometimes this script
-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Frank @ Impact
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Frank @ Impact
action application=bridge
I also tried to add this
action application=set data=hangup_after_bridge=false/
to keep the dialplan process on a-leg hangup. But that did not work
either.
Svn 10960 is what I am testing.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Can this command be used to run a bash script?
I wanted to do some sox processing on some recordings after the bridge
ends and thought I should use this command. But would like to do it in
bash.
Is there a better way?
If this is the right way, what is the syntax for calling the bash
Michael,
Got it working. Just a little simpler then you outlined.
I just added to my xml dialplan this line.
action application=set data=execute_on_answer=lua uuid_send_dtmf.lua
${uuid} 20 123/
I added this just before the bridge application.
I did this instead of adding an extra extension to
Pretty simple actually...
?xml version=1.0?
document type=freeswitch/xml
section name=dialplan description=Regex/XML Dialplan
context name=default
extension name=myextension
condition
action application=enum data=1551212/
action application=set
Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
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Freeswitch-users@lists.freeswitch.org
: [...@]time group_name command_string
/b
On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call
detects DTMF since
it's dual frequencies, rather tone_detect detects single frequencies
like fax tones.
I would just run an IVR with a session.read or session.getDigits to
collect DTMF.
Dan
On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact [EMAIL PROTECTED]
wrote:
Same thing with version 10640 build
How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)?
We have a carrier that penalizes us with a surcharge for short duration
calls (sound familiar?).
So when a FS caller (not a call center or predictive dialer) calls a
cell phone and gets a ring tone or calls
. But it is probably a better use
of time to approach this as a business issue.
My 2 cents.
On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)?
We have a carrier that penalizes us with a surcharge for short duration
current or install current svn on a different box.
/b
On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
Ideas? Am I doing something stupid or is tone_detect not just right
here?
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Freeswitch-users mailing list
Freeswitch-users
Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?
___
Freeswitch-users
Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?
-Frank
___
Freeswitch-users
.
-Original Message-
From: [EMAIL PROTECTED] [mailto:freeswitch-
What would need to happen after the tone is sent back out? Also, would
this be part of something like an IVR?
-MC
On Dec 5, 2008, at 7:22 AM, Frank @ Impact [EMAIL PROTECTED]
wrote:
Is there any dialplan instructions that could
Yes. listen in for 1 DTMF during a call and then signal back a
different DTMF.
-Original Message-
From: [EMAIL PROTECTED] [mailto:freeswitch-
So receive DTMF respond with more DTMF?
/b
On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:
After the tone is sent back out, we are done
and we will see what we can come up with.
-MC
On Dec 5, 2008, at 8:00 AM, Frank @ Impact [EMAIL PROTECTED]
wrote:
After the tone is sent back out, we are done. There is nothing left
to
do.
No, this key press detection is during a bridged call between two
parties. No IVR here. So
a pretty old rev. Any chance you could make current?
-MC
Sent from my iPhone
On Dec 5, 2008, at 5:09 PM, Frank @ Impact [EMAIL PROTECTED]
wrote:
I tried your suggested test. Here is the business end of the
extension
I tried.
action application=set data=DTMF1=false/
action
? If you call with a 16khz codec like g722 it should
choose 16k
On Fri, Oct 31, 2008 at 4:08 PM, Frank @ Impact [EMAIL PROTECTED]
wrote:
I have a cepstral Allison voice for 16khz. I am using javasrcipt
session.speak.
The audio output is not nearly as good as what comes out directly from
swift.
I am
I am trying to catch a key being pressed during a bridged call. The key
could be pressed by either leg of the call. When the key is pressed, I want
to play into both channels some sound file or send in some TTS output. Then
after the playback is done, allow the callers to resume their
Running FreeSwitch Version 1.0.trunk (9111) on RH 8
Record_session works fine with this
action application=set data=RECORD_ANSWER_REQ=true/
action application=set data=RECORD_STEREO=true/
action application=record_session
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