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 on
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
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]: 2009-11-
"GREAT SCOTT!!! Cannot execute batched statements!
If you are using mysql, make sure you are using MYODBC 3.51.18 or higher
and enable FLAG_MULTI_STATEMENTS"
I realize a bit off of list topic.
But I do have mysql 3.51.18 and higher but for the life of me , I cannot
seem to get the DSN config se
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
wrote:
Is there a way to bind a particular extension to a different dialplan
php and a different cdr php script than the
FreeSWITCH Version 1.0.trunk (12790M)
I have this in my DP
works fine as long as the call is long enough. But if the call is only,
say, 3-4 seconds long (or something very short like that), then the wav
file is never created with the audio in it.
Is there a w
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
http://lists.freeswitch.org/mailman/listinfo/freeswitch-
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 example,
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
: 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 and is
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 `/usr/
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: Thursda
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
- W
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: freeswitch-users@lists.freeswitch.o
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
[mailto:freeswitch-users-boun...@lists.
Maybe I am going about this all wrong. All I am trying to do is process
a recording file of a session after either one of the legs hangs up and
the call is over. I am just trying to convert the wav to mp3 and email
it off. So I have a bash script to do this. The dialplan is simple
enought
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 f
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 back
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
so by the time the lua is called, someone has hungup one of the legs.
In the lua script I am using this to try to end the record session to
the wav file so it g
I also tried to add this
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
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerr
I also tried this without success. This will not fire at all regardless
of who hangs up.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Frank @ Impact
-Original Message-
From
...@lists.freeswitch.org] On Behalf Of
Frank @ Impact
The problem I am seeing is that sometimes this script gets run and
sometimes it does not. I think it has to do maybe with which end hangs
up the phone. But I cannot seem to nail it down just yet
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,
The problem I am seeing is that sometimes this script gets run and
sometimes it does not. I think it has to do maybe with which e
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 script
Pretty simple actually...
BTW, this darn tone_detect is something I never could get working. It
did not matter which side I sent the tone from, it never got trapped by
my test he
Michael,
Got it working. Just a little simpler then you outlined.
I just added to my xml dialplan this line.
I added this just before the bridge application.
I did this instead of adding an extra extension to transfer to on
answer. Everything worked well. The DTMF was played to the calling
: [...@]
/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
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 mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/li
you were doing before I go any further.
-MC
On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <[EMAIL PROTECTED]>
wrote:
> On our last bill, the carrier said we had 27% short duration calls
(maybe
> they are wrong but it was on the bill). It is definitely not call
center.
> But these
On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <[EMAIL PROTECTED]>
wrote:
> On our last bill, the carrier said we had 27% short duration calls
(maybe
> they are wrong but it was on the bill). It is definitely not call
center.
> But these callers hangup as soon as they hear answer
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 shor
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
e_detect 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
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
Freeswi
7;s 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
I tried your suggested test. Here is the business end of the extension
I tried.
but I always got DTMF1=false in the info dump.
I am using FS 9210
I have tried sending a call from my sip phone connected to an asterisk
server to FS
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
ee 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
> pa
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 d
.
-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 dial
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
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 maili
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 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 guessing that speak is down sampling to 8khz. Is this true?
Is there a way to get speak to not do this?
Thanks
__
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 conversati
Running FreeSwitch Version 1.0.trunk (9111) on RH 8
Record_session works fine with this
But the sample rate and resulting wav file are way more information than I
need. I tried to half the sample rate with this inserted before we run
record_session.
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