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 thorou
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 wrote:
> async?
>
> On Nov 24, 2009, at 2:22 AM, velusamy velu
Yes, I am using async mode only..
On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris 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 playback_t
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 no
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:
>
>
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 'COMM
* 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 argv[2]=vf
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
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
pr
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
_
On Tue, Sep 15, 2009 at 3:37 PM, Alberto Escudero 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 with Freeswit
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
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 receiv
you would have to write a module in c that hooks the dtmf and throws
it away. This would work similar to how bind_meta works. That being
said, this is all much easier if your using 2833 as you don't need to
go to the extra cpu of detecting tones.
Mike
On Aug 8, 2009, at 9:10 AM, Frank @
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,
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 possi
stake with a huge effect. But thanks for all you help. :-)
Greetz
- Ursprüngliche Mail -
Von: "Brian West"
An: freeswitch-users@lists.freeswitch.org
Gesendet: Montag, 8. Juni 2009 17:12:35 GMT +01:00
Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] DT
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
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 Denert wrote:
> Hello!
>
> Is there a possibility to "detect" or "scan" whic
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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Andy 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 the logs you qu
that helps.
Cheers Andy
-Original Message-
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 thro
Andy wrote:
> It's not that digits get dropped some calls semm to handle dtmf perfectly
> and others don't seem to get dtmf at all.
>
> Can anyone shed any light opn this or suggest any solutions?
I can't help, but you could make it a lot easier for others to help you by
including the necessary
Hi,
I have an urgent issue if anyone can help. I have been running freeswitch
for 3-4 weeks now without issue. In the last 2 days some of the calls coming
into the switch seem to get set up in such a way that means they cannot
carry DTMF. ie on that call, no dtmf signals come through from the ph
Rupa Schomaker 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 Web
interfac
Also, in general, I believe you want the jitter buffers on the end-point
devices only. Not the guy in the middle. So, jitter buffer should be
enabled on the phone, not within FS -- unless FS is the endpoint (eg: IVR).
On Fri, May 8, 2009 at 7:05 AM, Rupa Schomaker wrote:
> Sound bugish to me -
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 wrote:
> Sorry for all the e-mail...
>
> If I turn off the jitter b
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.
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As a matter of interest, the other end (as reported in its SDP) is BroadWorks.
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Jason White 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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Freeswit
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 als
Anthony Minessale 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 external restart reload
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 wrote:
> Remko Kloosterman wrote:
> >
> >
> > Did you make a wireshark trace yet? You should be able to find out
> > exactly what's
Remko Kloosterman 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 problem to the on
g] Namens Jay Austad
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 any
009 19:57
To: freeswitch-users@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 the
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
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
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 just
>> Did you provide the menu you are using and what you expect to happen?
Here's the setup;
Caller -> FlowRoute - > FreeSwitch
You should file the bug with the guy who dreamed up RFC2833 ;)
Did you provide the menu you are using and what you expect to happen?
There are cases where the way you set it up could cause your problems.
You also have to realize that dtmf over sip is one of the top 10 gripes ppl
have with the pro
Right and that is the fix for this. If you have the sleep's in your
phrase macro's remove them and use the pause= param... you shouldn't
have any problems.
/b
On Mar 27, 2009, at 2:01 PM, Chris Fowler wrote:
I re-did the macros; the only change I could detect was the
elimination
of th
>>
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 elim
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 i
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)
>> 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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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
> play_and_co
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:
Trace:
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf(
Check out the bind_meta_app that exists in the default examples... I
think thats what you want.
/b
On Mar 17, 2009, at 4:13 PM, Cristian Talle wrote:
> 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
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
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 a
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 carri
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, becaus
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
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 ca
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 sol
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 wrot
we are not using inband tones. we are using rfc2833.
is it still neccessary, to do some extra programming? if yes: isn't
there a way for fs to recognize, that there is a rfc2833 and simply
does not play it back for the others?
2009/2/9 Anthony Minessale :
> 1) don't use inband tones for dtmf.
>
On Mon, Feb 9, 2009 at 1:34 PM, Nik Middleton
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:
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Capturing_RTP_With_tshark_.28Advanced.29
B
chael 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
wrote:
> Further to this message, DTMF works with PMCU but not with PMCA which
is the
> native format f
On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton
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 vs. what is actually being receiv
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 fo
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
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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 wrote:
> hi,
>
> i am having a smal
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
other
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
gt;
> Thanks,
> Klaus.
> Original-Nachricht
> > Datum: Tue, 27 Jan 2009 23:15:02 -0600
> > Von: Brian West
> > An: freeswitch-users@lists.freeswitch.org
> > Betreff: Re: [Freeswitch-users] DTMF with Early Media Disabled
>
> > If the dtmf is in the
n: 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 earl
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,
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
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 cal
> > 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 vers
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
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 :
...
...
As codecs g711 ULAW (PCMU):
in vars.xml.conf :
So i guess that using latest version with a few changes in your config
should wo
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
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,
The send dtmf is working. thanks
--
Thank you with regards,
Gopal,
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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 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
day, November 20, 2008 4:27 AM
Para: freeswitch-users@lists.freeswitch.org
Asunto: Re: [Freeswitch-users] DTMF
Hi,
I have added the line in mod_commands.c file and compiled the file by make
and make install
After that i have tested the barging. i get the same error
api originate sofia/internal/1000%172.20.176
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,
>
>
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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And also forgot to say one thing, I am using event socket.
--
Thank you with regards,
Gopal,
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>
> Is there any dtmf setting that needs to be changed in the eyebeam phone?
>
--
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Gopal,
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I dont understand, can you please brief me?
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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.
__
Hi,
I am using the event socket in freeswitch with audiocodes, and the client
as a softphone.
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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 8:3
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?
--
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Gopal,
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viado el: Monday, November 17, 2008 5:59 AM
Para: freeswitch-users@lists.freeswitch.org
Asunto: Re: [Freeswitch-users] DTMF
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
These aren't inserting 1003 as the caller_id_number are they?
/b
On Nov 18, 2008, at 5:19 AM, Baskar wrote:
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>
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Hi,
I am working on Dtmf signal on barging
I have added these line to default.xml
i eavesdrop on extension 1003 by dialing 881003
1003 is connected to mobile no
but when i dial 881003 it get hangup
i hav
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 manual
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
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 *2
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 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
== 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
I just want to know what is what is the value to pass in
that parameter
Thanks in a
Hi,
I want to pass the DTMF digits through api command
i find the api command *api uuid_send_dtmf* <*uuid*>
I just want to know what is what is the value to pass in that
parameter
Thanks in advance
--
Warm Regards,
N.Baskar
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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
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