Hi All,
Thought I would share my solution to this DTMF problem: it turns out my
ISP was capping my bandwidth dropping packets to keep the connection
1Mbps, so the experienced DTMF loss was actually packets being discarded.
On my way to this discovery I tested Freeswitch DTMF quite
Your not telling anything to call your callback.
On Nov 24, 2009, at 1:03 AM, Baskar wrote:
Hi,
I want to check value given to the javascript with conditions whether it is
voicefile, extension or mobile Number when i press the dtmf value.
Steps i need to check in javascript:
When i
async?
On Nov 24, 2009, at 2:22 AM, velusamy velu wrote:
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set playback_terminators to cut the
playback when giving the digits. I have faced problem that DTMF event has not
Yes, I am using async mode only..
On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote:
async?
On Nov 24, 2009, at 2:22 AM, velusamy velu wrote:
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set
1. can you supply a trace of this esl communications.
2. is it inband or rfc2833 dtmf ?
MIke
On Nov 24, 2009, at 3:59 AM, velusamy velu wrote:
Yes, I am using async mode only..
On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote:
async?
On Nov 24, 2009, at 2:22 AM,
* Hi,*
*
*
*I want to check value given to the javascript with conditions whether it is
voicefile, extension or mobile Number when i press the dtmf value.*
*
*
*Steps i need to check in javascript:*
*
*
*When i Press the DTMF value 1 it should check the 3 condition*
*
*
If the Value for
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set playback_terminators to cut the
playback when giving the digits. I have faced problem that DTMF event has
not come if DTMF given while playing voice files. I have received
Hi All,
I have an issue that when my call volumes on my FS IVR box 30 calls DTMF
digits are lost (using RFC2833). It is definitely load related as it all
works perfectly under 30 calls.
Any pointers or a solution to the problem?
Thanks,
Michael
That's a pretty small problem description to be so sure about something.
It would probably be better to capture some evidence of the exact problem
you are having since we are using computers and we need to see the computers
in action doing something specifically incorrect to diagnose any sort of
Hi Anthony,
Thanks for the input. I will try reproduce the problem give you
something more concrete to work with log it in Jira.
Thanks again,
Michael
On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
That's a pretty small problem description to be
After digging into this issue, it might the case that the implementation
of out-bound DTMF of the client i am using does not properly increments
CSeq per DTMF.
For those interested, i am currently integrating OpenBTS with Freeswitch! :)
-aep
--
Stopping junk mailers is good for the environment
On Tue, Sep 15, 2009 at 3:37 PM, Alberto Escudero aep.li...@it46.se wrote:
After digging into this issue, it might the case that the implementation
of out-bound DTMF of the client i am using does not properly increments
CSeq per DTMF.
For those interested, i am currently integrating OpenBTS
Hi,
I am using the function session.collectInput and session.streamFile to
collect a number of DTMF digits.
If the DTMF digits are sent in the RTP, i can collect several digits until
timeout. No problem there! If the DTMFs are received as a sequence of SIP
INFO packages, collectInput only
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
Hello,
If I wanted a bridged call to a gateway to use inband DTMF for
incoming recognition and outgoing generation I'm unclear on what to do
because the wiki clearly states[1] not to use the start_dtmf and
start_dtmf_generate together for cause of loops.
Wouldn't it be technically possible
:35 GMT +01:00
Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] DTMF Problems
I wrote that to demonstrate that exact situation but you still can't tell if
they are inband or info :P
/b
On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote:
Rudolf,
I believe
Hello!
Is there a possibility to detect or scan which DTMF mode is sent by the
calling CPE so that I can establish logical interrogation in my configuration?
Greetz
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Rudolf,
I believe there is a snippet in the sample XML dialplan to detect
the lack of telephone-event in the SDP and activate inband detection.
You could use that for inspiration.
On Mon, Jun 8, 2009 at 2:42 AM, Rudolf Denertrden...@tng.de wrote:
Hello!
Is there a possibility to detect or
I wrote that to demonstrate that exact situation but you still can't
tell if they are inband or info :P
/b
On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote:
Rudolf,
I believe there is a snippet in the sample XML dialplan to detect
the lack of telephone-event in the SDP and activate
-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason
White
Sent: 15 May 2009 08:47
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF not comming through on some calls
Andy a...@fabulous4
Andy a...@fabulous4.co.uk wrote:
The DTMF method was efault which I believe is info but I've now set it
explicitly to rfc2833 inband to see if that helps. Is there a way I can tell
from the logs that this is the case and that my config changes have worked.
This is in the logs, and (assuming
I've narrowed this problem down.
When I call my ISP's DTMF test and issue DTMF from the Snom phone, do_2833()
from switch_rtp.c is never called, as evidenced by freeswitch.log.
However, if I call a friend's FreeSWITCH box from the phone (via my FreeSWITCH
instance), do_2833() is called. It is
Jason White ja...@jasonjgw.net wrote:
It is also called if I use the voicemail
extension on my local FreeSWITCH.
Apologies for the nonsense - I meant that switch_rtp_dequeue_dtmf() is called
in that case, for DTMF detection.
___
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As a matter of interest, the other end (as reported in its SDP) is BroadWorks.
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Sorry for all the e-mail...
If I turn off the jitter buffer that I had set in the dialplan extension for
that provider, DTMF is correctly sent and detected by the other side.
I suspect a bug, but maybe this is the desired behaviour.
___
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much
documentation as you have so it can be researched and resolved.
On Fri, May 8, 2009 at 3:46 AM, Jason White ja...@jasonjgw.net wrote:
Sorry for all the e-mail...
If I turn
Rupa Schomaker r...@rupa.com wrote:
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much
documentation as you have so it can be researched and resolved.
If someone could add it to Jira, I'll detail the issue here. The Jira
Verzonden: woensdag 6 mei 2009 20:57
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] DTMF recognition flaky
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed anything. I'm using
voicepulse for my
Remko Kloosterman r.klooster...@mtel.nl wrote:
Did you make a wireshark trace yet? You should be able to find out
exactly what's going on there, which protocol is used, etc. We've had
our share of problems with DTMF over SIP trunks as well.
I've just discovered that I'm having a similar
you may have a sonus infection
try some of the stuff from here under DTMF
http://wiki.freeswitch.org/wiki/RTP_Issues
On Thu, May 7, 2009 at 5:16 AM, Jason White ja...@jasonjgw.net wrote:
Remko Kloosterman r.klooster...@mtel.nl wrote:
Did you make a wireshark trace yet? You should be
Anthony Minessale anthony.miness...@gmail.com wrote:
you may have a sonus infection
try some of the stuff from here under DTMF
http://wiki.freeswitch.org/wiki/RTP_Issues
Thank you for the suggestion.
I tried both the Sonus and Cisco settings in the external profile (running
sofia profile
Using the default installation, I've noticed that when I (or someone
else) calls in on my SIP trunk and keys in an extension, not all of
the numbers are recognized unless they hold the key down for at least
1/2 second to a second.
Is there a way to improve DTMF recognition so people can
Well it depends.. first off are you doing inband dtmf or RFC2833?
Secondly what SVN rev are you running?
/b
On May 6, 2009, at 1:44 PM, Jay Austad wrote:
Using the default installation, I've noticed that when I (or someone
else) calls in on my SIP trunk and keys in an extension, not all of
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed anything. I'm using
voicepulse for my SIP trunks. Is there an option I can add to that
definition to force RFC2833?
--
jay austad | 612.423.1433 | aus...@signal15.com
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF recognition flaky
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed anything. I'm using
voicepulse for my SIP trunks. Is there an option I can add to that
definition to force
Sent: Wednesday, March 25, 2009 12:43
btw you'll have to reinstall your phrase macros make vm-sync I
think should do it if it doesn't let me know... we removed the 250ms
sleeps and that was the problem which we fixed.
I re-did the macros; the only change I could detect was the
Did you provide the menu you are using and what you expect to happen?
Here's the setup;
Caller - FlowRoute - FreeSwitch
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
invalid-sound=ivr/ivr-that_was_an_invalid_entry.wav
Any thoughts on why FS saw all digits 1029 but only reports '029'?
2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect()
digits '029'
Config:
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
First off what SVN rev? Remember when reporting issues try to
include all the information you can!
/b
On Mar 25, 2009, at 1:19 PM, Chris Fowler wrote:
Any thoughts on why FS saw all digits 1029 but only reports '029'?
2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364
First off what SVN rev? Remember when reporting issues try to include all
the information you can!
Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647)
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Please review this link http://wiki.freeswitch.org/wiki/Reporting_Bugs
The rules are try to reproduce this on SVN Trunk... I am pretty sure
we fixed this one already.
/b
On Mar 25, 2009, at 1:49 PM, Chris Fowler wrote:
Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647)
btw you'll have to reinstall your phrase macros make vm-sync I
think should do it if it doesn't let me know... we removed the 250ms
sleeps and that was the problem which we fixed.
/b
On Mar 25, 2009, at 1:49 PM, Chris Fowler wrote:
First off what SVN rev? Remember when reporting
Hi,
Is there any easy way to get in FS the same behavior as when using the
d flag with asterisk's Dial command?
I need FS to jump to a different extension if the caller presses a digit
while waiting for the called party to answer.
*...d*: intercepts any dtmf while waiting for the call to be
that is interesting. we are receiving the dtmf digits over 2833. might
it be possible, that we receive 2833 AND inband (we asked our carrier
for 2833, because we had problems with inband and fs - and we got it)?
is there something we can setup in fs or is it a problem wich only our
carrier can
Well if they are sending both they are broken. I would call and yell
at them and beat them with a cluebat.
/b
On Feb 11, 2009, at 10:42 AM, Dennis wrote:
that is interesting. we are receiving the dtmf digits over 2833. might
it be possible, that we receive 2833 AND inband (we asked our
i can't tell, if they are sending both, but it seems so. we get 2833
for sure. they were kind enough to give it to us, because inband seems
to be quite unreliable over sip.
how can in find out, if both are coming and is there a way to block
inband to test?
perhaps we need both: if we bridge an
turn on the start_dtmf app and dial digits from the outside.. if you
get duplicate digits then they are sending both.
/b
On Feb 11, 2009, at 11:14 AM, Dennis wrote:
i can't tell, if they are sending both, but it seems so. we get 2833
for sure. they were kind enough to give it to us, because
ok, i will try this, but how can it be possible, that inband tones are
audible in conference, when we do not even have start_dtmf activated?
i just don't understand, why it must be dtmf inband, if the tones are
audible and how they can be audible, if start_dtmf is not set.
is it, because the
On Feb 11, 2009, at 12:23 PM, Dennis wrote:
ok, i will try this, but how can it be possible, that inband tones are
audible in conference, when we do not even have start_dtmf activated?
They aren't really sending 2833.
i just don't understand, why it must be dtmf inband, if the tones are
If your in a conference and your hearing other people hitting dtmf
digits that IS inband, it means that the place upstream that is doing
inband to 2833 conversion is not properly clipping the dtmf, this
probably needs to be fixed on that device.
Mike
On Feb 10, 2009, at 9:58 AM, Dennis
hi,
i am having a small problem with the dtmf-sounds...
if i press a dtmf digit while i am bridged with another leg, the other
side will hear the dtmf sound.
this is very annoying and even worse in a conference, when multiple
people can press dtmf digits (for (un-)muting themselves or using
1) don't use inband tones for dtmf.
2) post a bounty to have FS clip the audio for x milliseconds when a tone is
detected. (you will still hear faint clicks between the start of the tone
and when the clipping activates)
On Mon, Feb 9, 2009 at 8:59 AM, Dennis oderm...@googlemail.com wrote:
hi,
Hi Guys,
I have an IVR that's working fine on internal extensions, but when a
call is via my sip GW, they're not being trapped.
I have tried the following in the gw profile
param name=dtmf-type value=rfc2833/
param name=rfc2833-pt value=101/
param name=pass-rfc2833 value=false/
I
Middleton
Sent: 09 February 2009 20:10
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] DTMF not being recognised
Hi Guys,
I have an IVR that's working fine on internal extensions, but when a
call is via my sip GW, they're not being trapped.
I have tried the following
On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Further to this message, DTMF works with PMCU but not with PMCA which is the
native format for this sip provider.
Any chance you could get some debug information? I'm wondering what is
actually being sent
Collins
Sent: 09 February 2009 21:27
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF not being recognised
On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Further to this message, DTMF works with PMCU but not with PMCA which
On Mon, Feb 9, 2009 at 1:34 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Forgive me, I'm not sure how I get that info with FS, can you enlighten
me?
I was thinking of something like Wireshark. You can also check out this:
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF with Early Media Disabled
If the dtmf is in the media stream ie 2833 and you can't establish
media then no you wouldn't. Have you tried to do a pre_answer
instead of an answer to establish early media?
/b
On Jan 27
-Nachricht
Datum: Tue, 27 Jan 2009 23:15:02 -0600
Von: Brian West br...@freeswitch.org
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF with Early Media Disabled
If the dtmf is in the media stream ie 2833 and you can't establish
media then no you
Klaus Teller napsal(a):
I know it works perfectly when pre_answer is called. That is, when early
media is activated. I was just trying to figure out what is the expected
behavior when pre_answer is not called.
I want to get DTMF from users without having them billed by their carriers.
Hi,
My settings does not allow me to test the following right now. So I'm wondering
if somebody knowledgeable could help me answer the following question.
I do know that if i call Freeswitch, i can use Javascript to read DTMF even
without answering the call. My question is can i do this even
If the dtmf is in the media stream ie 2833 and you can't establish
media then no you wouldn't. Have you tried to do a pre_answer
instead of an answer to establish early media?
/b
On Jan 27, 2009, at 11:04 PM, Klaus Teller wrote:
Hi,
My settings does not allow me to test the following
Hi,
I'm using freeswitch to receive incoming calls from a sip provider namely
AQL. When my freeswitch box is connected directly to the internet everything
works fine. When I place a firewall/router inbetween the box and the
internet, the software registers with the sip provider ok and answers
Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is
I had some issues with some previous versions of FS , in trunk looks
that is fixed. ( Notice current svn revision is 10609 )
in sip profiles i have :
...
param name=rfc2833-pt value=101/
param name=dtmf-duration value=100/
param name=codec-prefs value=$${global_codec_prefs}/
param
On Dec 5, 2008, at 6:08 AM, Jan Kubr wrote:
Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
So i guess that using latest version with
Hi,
The send dtmf is working. thanks
--
Thank you with regards,
Gopal,
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Hi,
Thanks for the support from *Brian West, Michael S Collins,Birgit Arkesteijn,
Cesar Cepeda, Michael Jerris, Gopala krishnan*.
DTMF is working fine in barging and Conference.
--
Warm Regards,
N.Baskar
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Hi cesar,
If i have added these line in mod_commands.c
stream-write_function(stream,+OK\n); just after inserting the DTMF
before the goto done;
When i compile by command *make* it get these error
*Compiling mod_commands.c...
mod_commands.c: In function âunload_functionâ:
mod_commands.c:869:
those errors are not caused by that change, either you updated only
parts of the code (that module maybe) and didn't update the rest of
FreeSWITCH or you have a merge conflict or other change in that file.
Mike
On Nov 21, 2008, at 9:59 AM, Baskar wrote:
Hi cesar,
If i have added these
Hi,
I was trying this dtmf stuff for me also its not working. whenever i used
to send the dtmf you know i get a beep. whats wrong?
--
Thank you with regards,
Gopal,
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you have to remember that just because you send DTMF to a phone via
RTP or SIP INFO the phone doesn't have to render them. The best way
to test this is with an ATA since it will render the tones most
likely. Many ip phones do NOT render the tones to the speaker.
/b
On Nov 20, 2008, at
Hi,
I am using the event socket in freeswitch with audiocodes, and the client
as a softphone.
--
Thank you with regards,
Gopal,
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Your phone must not be rendering them. I just tested this and its
working fine. X-Lite/eyeBeam
/b
On Nov 20, 2008, at 8:55 AM, Gopala krishnan wrote:
Hi,
I am using the event socket in freeswitch with audiocodes, and the
client as a softphone.
I dont understand, can you please brief me?
--
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Gopal,
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Is there any dtmf setting that needs to be changed in the eyebeam phone?
--
Thank you with regards,
Gopal,
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It worked by default on mine... I'm on the Mac version of eyeBeam.
/b
On Nov 20, 2008, at 9:03 AM, Gopala krishnan wrote:
Is there any dtmf setting that needs to be changed in the eyebeam
phone?
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Doesn't matter.. the api call is the same via event socket or cli.
(as in they call the exact same code with NO differences)
/b
On Nov 20, 2008, at 9:07 AM, Gopala krishnan wrote:
And also forgot to say one thing, I am using event socket.
--
Thank you with regards,
Gopal,
These aren't inserting 1003 as the caller_id_number are they?
/b
On Nov 18, 2008, at 5:19 AM, Baskar wrote:
action application=db data=insert/spymap/$
{caller_id_number}/${uuid}/
action application=db data=insert/last_dial/$
{caller_id_number}/${destination_number}/
Hi,
I want to pass the DTMF digits through api command
i find the api command *api uuid_send_dtmf* *uuid* dtmf_data
I just want to know what is dtmf_data what is the value to pass in that
parameter
Thanks in advance
--
Warm Regards,
N.Baskar
___
dtmf_data == The digits you wish to pass.
Tip... try then ask ;)
/b
On Nov 17, 2008, at 5:33 AM, Baskar wrote:
Hi,
I want to pass the DTMF digits through api command
i find the api command
api uuid_send_dtmf uuid dtmf_data
I just want to know what is dtmf_data what is the value to pass in
Hi Baskar,
I assume the dtmf_data is a string of one or more dtmf digits. So for
example:
1 or *123#
Why don't you try it on the console and see what you get?
(Please anyone correct me if I'm wrong.)
Cheers, Birgit
On 17/11/08 11:33, Baskar wrote:
Hi,
I want to pass the DTMF digits
On Nov 17, 2008, at 5:44 AM, Birgit Arkesteijn wrote:
Why don't you try it on the console and see what you get?
You're right... and this is good advice... TRY then Ask ;) Things are
simple most of the time ;)
(Please anyone correct me if I'm wrong.)
Cheers, Birgit
Hi,
i have tried it before itself first i pass one digit
api uuid_send_dtmf c08f77be-fbed-44c3-a2a7-8650d88b0e33 *2 *
*
output:*
Content-Type: api/response
Content-Length: 14
-ERR no reply
Then i passed all the values in the barging
api uuid_send_dtmf baf82956-111d-4cd8-9568-47010ac8bd20
Hi Brain,
I am working on DTMF signals during *eavesdrop* and in* CONFERENCE *
DTMF signal is *not working* through* even socket api command *
I tried in conference also when we manually done in softphone it work . when
i press the # button it hangup and * for mute etc. it works fine but when
On Nov 17, 2008, at 9:18 PM, Baskar [EMAIL PROTECTED] wrote:
Hi Brain,
Hey, the guy is smart but his name ain't Brain!
I am working on DTMF signals during eavesdrop and in CONFERENCE
DTMF signal is not working through even socket api command
I tried in conference also when we
Hi,
In barging if we want to pass the DTMF signals. For example in barging
- 2 to speak with the uuid
- 1 to speak with the other half
- 3 to engage a three way
- 0 to restore eavesdrop.
- * to next channel.
I want pass these DTMF signals through event socket api uuid_send_dtmf
Hi,
I'm calling a registered soft phone (ext. 1003) via the event socket interface.
That is, on one side i have some Java code connecting to the Freeswitch event
socket interface and placing calls and on the other hand i have the soft phone
registered to Freeswitch and awaiting for calls.
you should be looking for the DTMF event and not reacting to any others
Event-Name: DTMF
any other ones are not necessarily related to what you want.
On Mon, Oct 27, 2008 at 8:49 AM, Klaus Teller [EMAIL PROTECTED] wrote:
Hi,
I'm calling a registered soft phone (ext. 1003) via the event
(CS_EXCHANGE_MEDIA or
CS_EXECUTE state). But then, DTMF-star doesn't always have these two states.
Klaus.
Original-Nachricht
Datum: Mon, 27 Oct 2008 11:19:18 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users
or CS_EXECUTE state). But then, DTMF-star doesn't always
have these two states.
Klaus.
Original-Nachricht
Datum: Mon, 27 Oct 2008 11:19:18 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Star
Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
if this is a bridged call you will get one on each leg as the dtmf passes
from one leg to the other.
if in some cases the dtmf is intercepted by something like the
bind_meta_app
then you may only see 1.
On Mon, Oct 27, 2008
]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
I'm not an authority on this, but I have spotted some things that might
help
you figure this out
Your events show up with different unique-ids - Unique-ID:
34b83622-a473-11dd-8207-2b46fcff01af
Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
I'm not an authority on this, but I have spotted some things that might
help
you figure this out
Your events show up with different unique-ids - Unique-ID:
34b83622-a473-11dd-8207-2b46fcff01af
and Unique-ID: 34adac7a-a473
14:04:14 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
there are 2 channels there?
one is ulaw and the other is g722 they are both getting a dtmf event?
also this output suggests older
Are you using inband dtmf anywhere in this mix?
/b
On Oct 27, 2008, at 3:13 PM, Klaus Teller wrote:
As far as i can tell, there is one single channel. Call is
initiated via the socket interface to the extension 1003 and parked.
Or does parking generate a second channel?
I'm using
that it's RFC 2833.
Klaus.
Original-Nachricht
Datum: Mon, 27 Oct 2008 15:17:52 -0500
Von: Brian West [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
Are you using inband dtmf anywhere in this mix?
/b
]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
Are you using inband dtmf anywhere in this mix?
/b
On Oct 27, 2008, at 3:13 PM, Klaus Teller wrote:
As far as i can tell, there is one single channel. Call is
initiated via
Klaus Teller wrote:
I tried the following but for unknown reason, the caller is not getting
anything:
JavaSession s = new JavaSession(uuid);
s.answer();
s.streamFile(/usr/local/freeswitch/sounds/1.wav);
s.execute(send_dtmf, [EMAIL PROTECTED]);
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