Re: [Freeswitch-users] DTMF Digits Lost when Under Load
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 thoroughly
never actually found any problems even at hundreds of concurrent calls.
Here is how I tested, who knows this might be useful to someone:
- I used SIPp to generate calls a Python script to log the received
DTMF digits
- SIPp command line:
- sipp -sf dtmfSenario.xml -d 1 -s 451 -l 96 -mp 5606 -i
xxx.xxx.xxx.xxx
- dtmfSenario.xml below
- Dialplan:
- extension name=test_dtmf_capture_test
!--Grab calls for dialing --
condition field=destination_number expression=(^100100$)
action application=answer/
action application=python data=writeDtmfStats/
/condition
/extension
- Python:
- import sys
from freeswitch import *
def get_number(session,invalid,num=20):
digits = session.getDigits(num, , 15000)
consoleLog(info,Got '%s' digits from user.\n % digits)
if digits == '':
# Invalid call
if invalid == 3:
consoleLog(info,Three invalid
attempts!!\n)
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav)
session.hangup()
sys.exit(0)
else:
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav)
get_number(session,invalid + 1)
else:
consoleLog(info,Got a valid number: %s,
proceeding...\n % digits)
return digits
def handler(session, args):
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav)
numberToDial = get_number(session,2,num=10)
consoleLog('info','Got 10 DTMF digits. Writing 1 to file...\n')
fo = open('/tmp/dtmfData.csv','a')
fo.write('1\n')
fo.close()
# Do some stuff wait for SIPP to hangup
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav)
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav)
return
- DTMF senario file:
- # cat dtmfSenario.xml
?xml version=1.0 encoding=ISO-8859-1?
!DOCTYPE scenario SYSTEM sipp.dtd
!-- This program is free software; you can redistribute it
and/or --
!-- modify it under the terms of the GNU General Public License
as --
!-- published by the Free Software Foundation; either version 2 of
the --
!-- License, or (at your option) any later
version.--
!--
--
!-- This program is distributed in the hope that it will be
useful,--
!-- but WITHOUT ANY WARRANTY; without even the implied warranty
of --
!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See
the --
!-- GNU General Public License for more
details. --
!--
--
!-- You should have received a copy of the GNU General Public
License --
!-- along with this program; if not, write to
the --
!-- Free Software Foundation,
Inc.,--
!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307
USA --
!--
--
!-- Sipp 'uac' scenario with pcap (rtp)
play --
!--
--
scenario name=UAC with media
!-- In client mode (sipp placing calls), the Call-ID MUST
be --
!-- generated by sipp. To do so, use [call_id]
keyword.--
send retrans=500
![CDATA[
INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp sip:s...@[local_ip]:[local_port];tag=[call_number]
To: sut sip:[servi...@[remote_ip]:[remote_port]
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 18 100
Re: [Freeswitch-users] DTMF javasript
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 Press the DTMF value 1 it should check the 3 condition
If the Value for argv[2]=vfsurya means it is a voice file so it should play
the Voice file
If the Value for argv[2]=1001 means it is a extension. The call should Bridge
the extension
If the Value for argv[2]=9841799874 means it is a Mobile number. The call
should Bridge the Mobile number
var exit = false;
var dtmf_digits = ;
var repeat = 0;
var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number
function onInput( session, type, data, arg )
{
if ( type == dtmf )
{
console_log( info, Got digit + data.digit + \n );
if ( data.digit == 1 )
{
if(argv[2].startswith(vf))
{
var voice2=voice.substring(2)+br /
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/+voice2+.wav,
onInput );
}
else if(argv[2].length==4)
{
console_log( info, Got voicefile + argv[2] + \n
);
session.execute(bridge,
sofia/internal/+argv[2]+%192.168.1.2, onInput );
}
else
{
session.execute(bridge,
sofia/default/sip:+argv[2]+@192.168.1.135:5066, onInput );
}
}
}
}
But if 1 is pressed there is no event trigger but it get the dtmf value as 1
in freeswitch console.
can any one specify what is the error or correct me where i am wrong.
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Re: [Freeswitch-users] DTMF Event is not coming while using playback terminators.
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 come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Event is not coming while using playback terminators.
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 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 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Event is not coming while using playback terminators.
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, 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 come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF javasript
* 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]=vfsurya means it is a voice file so it should play
the Voice file
*If the Value for argv[2]=1001 means it is a extension. The call should
Bridge the extension*
*If the Value for argv[2]=9841799874 means it is a Mobile number. The call
should Bridge the Mobile number*
*
*
*var exit = false;*
*var dtmf_digits = ;*
*var repeat = 0;*
*var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number*
*
*
*
*
*function onInput( session, type, data, arg ) *
*{*
* if ( type == dtmf ) *
* {*
*console_log( info, Got digit + data.digit + \n );*
*if ( data.digit == 1 ) *
* **{*
*if(argv[2].startswith(vf))*
* **{*
* **var voice2=voice.substring(2)+br /*
*
**session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/+voice2+.wav,
onInput );*
* **}*
* **else if(argv[2].length==4)*
* **{*
* **console_log( info, Got voicefile + argv[2] + \n );*
* **session.execute(bridge, sofia/internal/+argv[2]+%192.168.1.2,
onInput ); *
* **}*
* **else*
* **{*
* **session.execute(bridge, sofia/default/sip:+argv[2]+@
192.168.1.135:5066, onInput ); *
* **}*
*}*
*}*
*}*
*
*
*But if 1 is pressed there is no event trigger but it get the dtmf value as
1 in freeswitch console. *
*
*
*can any one specify what is the error or correct me where i am wrong.*
*
--
Thanks with Regards,
N.Baskar *
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[Freeswitch-users] DTMF Event is not coming while using playback terminators.
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 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? Thanks, Velusamy ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF Digits Lost when Under Load
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Digits Lost when Under Load
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 problem. Take the time to describe the origin and destination of your calls, the call flow, the hardware in use on both ends of the call, detailed console logs on debug level, (maybe even uncomment the 2833 debug ifded in switch_rtp.c) and gather something to go on besides I seem to be losing dtmf) maybe a packect capture of the networking interface on both ends of these calls. Also problems should be reported to http://jira.freeswitch.org not this mailing list. Save us a step if you report a jira and provide all the info above or we will just have to ask for it again. On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop micha...@voxcore.voxtelecom.co.za wrote: 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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Digits Lost when Under Load
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 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 problem. Take the time to describe the origin and destination of your calls, the call flow, the hardware in use on both ends of the call, detailed console logs on debug level, (maybe even uncomment the 2833 debug ifded in switch_rtp.c) and gather something to go on besides I seem to be losing dtmf) maybe a packect capture of the networking interface on both ends of these calls. Also problems should be reported to http://jira.freeswitch.org not this mailing list. Save us a step if you report a jira and provide all the info above or we will just have to ask for it again. On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop micha...@voxcore.voxtelecom.co.za wrote: 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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
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 receives the first one. Any ideas? -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
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 with Freeswitch! :) -aep We are very interested in seeing how this pans out. Please keep us posted on your progress and definitely come back when you have questions. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF
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 receives the first one. Any ideas? -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF disable a few secs after call starts
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, after 15 seconds, is there a way to get FS to stop passing DTMF events? Would I have to try to force asterisk to use rfc2833 when sending the call to FS and when accepting it back from FS? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF confusion
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 to generate DTMF only on the outbound leg and recognize DTMF only on the inbound leg without interference? I assume I'm not understanding something correctly here - could somebody elaborate? The end goal for me is to detect the absense of telephone-event rtpmap and enable inband DTMF from a gateway. Thanks! - Jesse [1] http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Problems
Hello, I can give the all-clear! It was my mistake ( ...and it was a silly one :-/ ) I had to applications that interfere each other. They are: !--action application=start_dtmf/ -- !--action application=start_dtmf_generate data=true/-- I don't know why I skip that in my dialplan!!! http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Yes! A small mistake with a huge effect. But thanks for all you help. :-) Greetz - Ursprüngliche Mail - Von: Brian West br...@freeswitch.org 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] 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 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. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF Problems
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Problems
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 scan which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? Greetz ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 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. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF not comming through on some calls
Apologies. The freeswitch software is receiving incoming calls from a voip gateway. I'm using voiptalk in the UK. 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. Most of the phones are mobiles but some landlines as well. I've done a detailed analysis of the logs and the calls that don't work are missing what appear to be critical actions in the debug. Namely: 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 And then a little later in the call 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp() Audio params changed for sofia/external/07540526...@194.145.190.143 from 194.145.190.143:11780 to 87.238.72.155:16968 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/07540526...@194.145.190.143] 77.86.49.249 port 21054 - 87.238.72.155 port 16968 codec: 8 ms: 20 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968] 2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state() Processing Reinvite 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526...@194.145.190.143 entering state [completed][200] 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526...@194.145.190.143 entering state [ready][200] These lines appear for calls that work and not when they don't. Hope 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 through on some calls Andy a...@fabulous4.co.uk 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 information with your question. For example, what DTMF method is configured in the SIP profiles - RFC2833 or Info, or are you using inband DTMF detection? What are the phones, and how are they connected to your FreeSWITCH system? What relevant information appears in your FreeSWITCH logs? For example, when debug-level logging is enabled, you should see log entries related to the DTMF detection. Check whether there are differences between the calls that work and those which don't. If it appears to be a bug, test whether you can reproduce it with the latest code taken from svn trunk. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF not comming through on some calls
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 the logs you quoted were taken after any relevant configuration change), they indicate that RFC2833 is indeed being used. This is also the default in the supplied Sofia profiles. Most of the phones are mobiles but some landlines as well. I've done a detailed analysis of the logs and the calls that don't work are missing what appear to be critical actions in the debug. Namely: 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 That's turning on RFC2833, as I understand it, for DTMF detection. And then a little later in the call 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp() Audio params changed for sofia/external/07540526...@194.145.190.143 from 194.145.190.143:11780 to 87.238.72.155:16968 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/07540526...@194.145.190.143] 77.86.49.249 port 21054 - 87.238.72.155 port 16968 codec: 8 ms: 20 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968] 2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state() Processing Reinvite 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526...@194.145.190.143 entering state [completed][200] 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526...@194.145.190.143 entering state [ready][200] These lines appear for calls that work and not when they don't. There are obviously SIP reinvite messages being received from your SIP provider, and which FreeSWITCH is processing successfully. What I'm wondering is whether your provider is always offering RFC2833, since, given the above, they seem to have complex call handling arrangements. For that, you would need to look at the SDP from the remote end in the calls for which DTMF isn't being detected properly. Fortunately, this is logged by FreeSWITCH when set to debug logging. What you're looking for is a line such as a=rtpmap:101 telephone-event/8000 If that isn't present, then something odd would appear to be going on at your SIP provider's end, which is what I personally suspect, since FreeSWITCH is correctly activating RFC2833 support on the channel in other cases. You can also obtain a sip trace: sofia profile external siptrace on which will show you exactly what you're receiving from your provider. Disclaimer: I'm not an expert, but I hope this helps anyway. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 also called if I use the voicemail extension on my local FreeSWITCH. Finally, if I call my ISP via PortAudio and use the pa dtmf command, do_2833() is called. It's either something in my configuration, or a bug. I'll keep looking. Anyone with ideas is welcome to offer suggestions. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
As a matter of interest, the other end (as reported in its SDP) is BroadWorks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 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. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 Web interface is a problem for me, and it doesn't seem to allow submissions by e-mail or in other ways. Basically, the problem is that RFC2833 DTMF isn't sent to the other side if a jitterbuffer is set in the dial plan extension for the outbound call with action application=set data=jitterbuffer_msec=180/ and the call originates from my SIP phone (a Snom 320). The FreeSWITCH logs show that do_2833() in switch_rtp.c isn't called in this case. I'll gladly provide further details if and when anyone has a chance to investigate, assuming that it isn't desired behaviour (which in my opinion it isn't). ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
Hi Jay, 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. Your problems could also be related to timing issues introduced by multiple gateways. Do you know some details on voicepulse's network? There's lots of variations in implementation out there, unfortunately not always fully compatible. Good luck, Remko Van: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] 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 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 On May 6, 2009, at 1:46 PM, Brian West wrote: 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 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 type in stuff without having to hold the keys down? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 problem to the one discussed in this thread. Here are the symptoms. 1. If I call FreeSWITCH from my Snom 320 SIP phone, DTMF recognition works perfectly. This is also true if I call a friend's FreeSWITCH system. 2. If I call a certain VoIP provider from the Snom phone, via FreeSWITCH (phone - FreeSWITCH - provider) and call the provider's DTMF test, DTMF recognition fails to work. Apparently this provider accepts only RFC-2833, which is what FreeSWITCH should be issuing - I haven't changed the settings in the external profile from the defaults. 3. If I call the same provider's DTMF test from PortAudio and issue the pa dtmf command, the provider recognizes the DTMF traffic correctly. I couldn't find any obvious configuration errors on the phone or in my internal and external Sofia profiles. I'll gladly run tshark if that's the next step to take. I can also try setting param name=pass-rfc2833 value=true/ in the internal profile, but this shouldn't be necessary, since as the wiki states in documenting this variable, FreeSWITCH should decode and re-encode the RFC2833 data anyway when this is set to false. I'll keep working on this, but in the meantime, suggestions are welcome. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 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 one discussed in this thread. Here are the symptoms. 1. If I call FreeSWITCH from my Snom 320 SIP phone, DTMF recognition works perfectly. This is also true if I call a friend's FreeSWITCH system. 2. If I call a certain VoIP provider from the Snom phone, via FreeSWITCH (phone - FreeSWITCH - provider) and call the provider's DTMF test, DTMF recognition fails to work. Apparently this provider accepts only RFC-2833, which is what FreeSWITCH should be issuing - I haven't changed the settings in the external profile from the defaults. 3. If I call the same provider's DTMF test from PortAudio and issue the pa dtmf command, the provider recognizes the DTMF traffic correctly. I couldn't find any obvious configuration errors on the phone or in my internal and external Sofia profiles. I'll gladly run tshark if that's the next step to take. I can also try setting param name=pass-rfc2833 value=true/ in the internal profile, but this shouldn't be necessary, since as the wiki states in documenting this variable, FreeSWITCH should decode and re-encode the RFC2833 data anyway when this is set to false. I'll keep working on this, but in the meantime, suggestions are welcome. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 external restart reloadxml after making the changes). This didn't help, unfortunately. If I were to make an informed guess, I would expect Cisco equipment to be at the other end, since my ISP has a strong relationship with Cisco. Whatever their solution for carriers is, this is likely to be it, but I could be wrong, of course. I find it interesting that dtmf over PortAudio works, but from the Snom phone it does not. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF recognition flaky
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 type in stuff without having to hold the keys down? -- jay austad | 612.423.1433 | aus...@signal15.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
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 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 type in stuff without having to hold the keys down? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 RFC2833? -- jay austad | 612.423.1433 | aus...@signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: 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 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 type in stuff without having to hold the keys down? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF recognition flaky
Hi Jay, Have to say my DTMF works flawlessly on thousands of calls. (SVN trunk from a couple of days ago. We handle around 100,000 calls/day via FS) That said, I've found it depends on your SIP trunk provider.That doesn't mean to say there isn't a problem; it's just that I haven't come across it. Know it's not helpful, but there you go. Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jay Austad Sent: 06 May 2009 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 there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | aus...@signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: 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 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 type in stuff without having to hold the keys down? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Missing Digits
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 elimination of the 250ms sleeps; and the change to: macro name=welcome pause=250 I'm running build 12782; should this have fixed it? If so, I will follow the bug reporting instructions you sent earlier. Thanks, Chris. Here's the errors caught today on my production system. 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '1101' 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '55' 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '015' 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '' 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Missing Digits
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
exit-sound=ivr/ivr-operator.wav
timeout =1
inter-digit-timeout=1500
max-failures=2
max-timeouts=7
digit-len=4
entry action=menu-exec-app digits=/^(10[0-2][0-9])$/
param=transfer $1 XML public/
entry action=menu-exec-app digits=/^(30\d{2})$/
param=transfer $1 XML default/
entry action=menu-exec-app digits=0 param=transfer 1000 XML
public/ !-- Send to the operator extension --
entry action=menu-exec-app digits=# param=transfer 6000 XML
default/
/menu
macro name=welcome pause=250
input pattern=(.*)
match
action function=play-file
data=/usr/local/freeswitch/sounds/fr1.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/fr2.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/if-u-know-ext-dial.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/fr3.wav/
/match
/input
/macro
macro name=top-menu pause=250
input pattern=(.*)
match
action function=play-file
data=/usr/local/freeswitch/sounds/if-u-know-ext-dial.wav/
action function=play-file
data=/usr/local/freeswitch/sounds/fr3.wav/
/match
/input
/macro
B: 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.
Still seeing multiple issues logged during ivr process for
mis-interpreted DTMF. Here's today's list from our production server.
2009-03-27 06:38:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
'1100'
2009-03-27 07:20:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
'1101'
2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '55'
2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '015'
2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input
''
2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 11:58:35 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '028'
2009-03-27 11:59:27 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '050'
2009-03-27 12:01:52 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 12:02:01 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000'
2009-03-27 12:02:41 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
2009-03-27 12:02:53 [DEBUG] switch_ivr_menu.c:548
switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006'
Any other debug I can capture to assist?
Thanks, Chris.
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[Freeswitch-users] DTMF Missing Digits
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
invalid-sound=ivr/ivr-that_was_an_invalid_entry.wav
exit-sound=voicemail/vm-goodbye.wav
timeout =1
inter-digit-timeout=1500
max-failures=3
max-timeouts=7
digit-len=4
entry action=menu-exec-app digits=/^(10[0-2][0-9])$/
param=transfer $1 XML public/
entry action=menu-exec-app digits=/^(30\d{2})$/
param=transfer $1 XML default/
entry action=menu-exec-app digits=0 param=transfer 1000 XML
public/
entry action=menu-exec-app digits=# param=transfer 6000 XML
default/
entry action=menu-top digits=9/
Trace:
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf()
RTP RECV DTMF 1:2000
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1301 do_2833() Send start
packet for [1] ts=1129880426 dur=160/160/2000 seq=2804
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=320/320/2000 seq=2805
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=480/480/2000 seq=2806
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=640/640/2000 seq=2807
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=800/800/2000 seq=2808
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=960/960/2000 seq=2809
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1120/1120/2000 seq=2810
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1280/1280/2000 seq=2811
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1440/1440/2000 seq=2812
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1600/1600/2000 seq=2813
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1760/1760/2000 seq=2814
2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [1] ts=1129880426 dur=1920/1920/2000 seq=2815
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [1] ts=1129880426 dur=2080/2080/2000 seq=2816
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [1] ts=1129880426 dur=2080/2080/2000 seq=2817
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [1] ts=1129880426 dur=2080/2080/2000 seq=2818
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf()
RTP RECV DTMF 0:2160
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start
packet for [0] ts=1129884426 dur=160/160/2160 seq=2819
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=320/320/2160 seq=2820
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=480/480/2160 seq=2821
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=640/640/2160 seq=2822
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=800/800/2160 seq=2823
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=960/960/2160 seq=2824
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1120/1120/2160 seq=2825
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1280/1280/2160 seq=2826
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1440/1440/2160 seq=2827
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1600/1600/2160 seq=2828
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1760/1760/2160 seq=2829
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=1920/1920/2160 seq=2830
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle
packet for [0] ts=1129884426 dur=2080/2080/2160 seq=2831
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [0] ts=1129884426 dur=2240/2240/2160 seq=2832
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [0] ts=1129884426 dur=2240/2240/2160 seq=2833
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet
for [0] ts=1129884426 dur=2240/2240/2160 seq=2834
2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf()
Re: [Freeswitch-users] DTMF Missing Digits
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_collect() digits '029' ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Missing Digits
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) ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Missing Digits
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) ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Missing Digits
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 issues try to include all the information you can! Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF detection during bridge
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 answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered... Thanks, Cristian ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 solve? 2009/2/10 Michael Jerris m...@jerris.com: 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. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 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 solve? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 inbound with another ivr on the outbound side, which is not sip and does not understand 2833, we need to pass inband through or something like this. or am i wrong with this? 2009/2/11 Brian West br...@freeswitch.org: 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 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 solve? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 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 inbound with another ivr on the outbound side, which is not sip and does not understand 2833, we need to pass inband through or something like this. or am i wrong with this? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 carrier just sends them as normal sound, which is played as a tone, without beeing used for dtmf? 2009/2/11 Brian West br...@freeswitch.org: 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 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 inbound with another ivr on the outbound side, which is not sip and does not understand 2833, we need to pass inband through or something like this. or am i wrong with this? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 audible and how they can be audible, if start_dtmf is not set. is it, because the carrier just sends them as normal sound, which is played as a tone, without beeing used for dtmf? I bet they don't know how to config their switch to do 2833. /b ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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 wrote: 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 anthony.miness...@gmail.com: 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, 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 functions). is there a way, to NOT let the other side hear the dtmf sound (but of course still make fs listening to it)? thanks for the help dennis ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF: Mute sound for the other side?
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 functions). is there a way, to NOT let the other side hear the dtmf sound (but of course still make fs listening to it)? thanks for the help dennis ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
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, 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 functions). is there a way, to NOT let the other side hear the dtmf sound (but of course still make fs listening to it)? thanks for the help dennis ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[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 in the gw profile param name=dtmf-type value=rfc2833/ param name=rfc2833-pt value=101/ param name=pass-rfc2833 value=false/ I should add that this sip provider works fine with asterisk. Anyone any ideas? Regards ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF not being recognised
Further to this message, DTMF works with PMCU but not with PMCA which is the native format for this sip provider. Regards From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik 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 in the gw profile param name=dtmf-type value=rfc2833/ param name=rfc2833-pt value=101/ param name=pass-rfc2833 value=false/ I should add that this sip provider works fine with asterisk. Anyone any ideas? Regards ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 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 received. A pcap at the far end to compare with a pcap at the near end would be quite enlightening. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF not being recognized
Forgive me, I'm not sure how I get that info with FS, can you enlighten me? DTMF also works with GSM and others, but not Alaw Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael 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 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 received. A pcap at the far end to compare with a pcap at the near end would be quite enlightening. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF not being recognized
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: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Capturing_RTP_With_tshark_.28Advanced.29 Being able to see what *actually* is going out over the wire (or coming in on the wire) can take much of the guess work out of debugging. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF with Early Media Disabled
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. I've heard that some carriers start billing as soon as early media is on. That's why i was wondering if DTMF (inband or out of band) can be received without answering or pre_answering. Thanks, Klaus. Original-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 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 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 early media is disabled on the inbound call? Thanks, Klaus. -- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF with Early Media Disabled
sorry, no, you can't do that. On Wed, Jan 28, 2009 at 7:55 AM, Klaus Teller klaus.tel...@gmx.net wrote: 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. I've heard that some carriers start billing as soon as early media is on. That's why i was wondering if DTMF (inband or out of band) can be received without answering or pre_answering. Thanks, Klaus. Original-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 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 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 early media is disabled on the inbound call? Thanks, Klaus. -- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF with Early Media Disabled
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. I've heard that some carriers start billing as soon as early media is on. Really? It violates the law in most Europian countries :-) I think you are from Germany - if such a carrier does it, the best you can do is to contact your Telco Regulator (Die Bundesnetzagentur - www.bundesnetzagentur.de) and ask for official correction. Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF with Early Media Disabled
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 early media is disabled on the inbound call? Thanks, Klaus. -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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, 2009, at 11:04 PM, Klaus Teller wrote: 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 early media is disabled on the inbound call? Thanks, Klaus. -- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF and firewall
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 calls but fails to respond to in call dtmf tones. AQL advised me to make sure I was using RFC2833 which I believe I have done by setting dtmf-type in my sip profile xml to 'RFC2833'. Can anyone advise me as to what other settings I should change to make the dtmf work correctly across the firewall/router? The router is currently set to allow all traffic. Many thanks for any help you can give. regards Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF from cell phones
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 recognized just fine, sometimes I had to wait for the file to be played etc. The important thing to note is that when using a SIP softphone (X-Lite) I have never had this problem, DTMF is recognized perfectly. So it's probably related to GSM or something. I was wondering whether anyone experienced the same and whether there is something I can do about it. There are a few DTMF-related variables in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt, dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with them a bit, but I don't really know what I'm doing.. Couldn't find any docs, either. Any ideas would be appreciated. Jan Kubr ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF from cell phones
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 name=use-rtp-timer value=true/
param name=rtp-timer-name value=none/
param name=inbound-codec-negotiation value=greedy/
...
As codecs g711 ULAW (PCMU):
in vars.xml.conf :
X-PRE-PROCESS cmd=set data=global_codec_prefs=PCMU,PCMA,GSM/
So i guess that using latest version with a few changes in your config
should work unless there's any other issue related to your sip provider
( PSTN / Media Gateway ), on this case you can get some captures of
sip/rtp traffic to check SDP and rtp Marks.
El vie, 05-12-2008 a las 12:08 +0100, Jan Kubr escribió:
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 recognized just fine, sometimes I had to wait for the
file to be played etc. The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
recognized perfectly. So it's probably related to GSM or something.
I was wondering whether anyone experienced the same and whether there
is something I can do about it. There are a few DTMF-related variables
in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
them a bit, but I don't really know what I'm doing.. Couldn't find any
docs, either.
Any ideas would be appreciated.
Jan Kubr
Cheers,
--
Angel Carpintero
ack ( at ) telefonica ( dot ) net
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Re: [Freeswitch-users] DTMF from cell phones
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 could not find any regularity in this, sometimes it is recognized just fine, sometimes I had to wait for the file to be played etc. The important thing to note is that when using a SIP softphone (X-Lite) I have never had this problem, DTMF is recognized perfectly. So it's probably related to GSM or something. I was wondering whether anyone experienced the same and whether there is something I can do about it. There are a few DTMF-related variables in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt, dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with them a bit, but I don't really know what I'm doing.. Couldn't find any docs, either. Any ideas would be appreciated. If it is coming from the sip provider as rfc 2833 dtmf, they are probably doing inband detection and failing at it. If you look at an rtp dump you can confirm this. If this is the case, there is nothing you can do on the FreeSWITCH side and the provider will have to fix it. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF from cell phones
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 a few changes in your config should work unless there's any other issue related to your sip provider ( PSTN / Media Gateway ), on this case you can get some captures of sip/rtp traffic to check SDP and rtp Marks. I tried trunk and the values for the variables (all except rtp-timer-name=none are already default in trunk), but only two things are different: 1. When I press a key, the read app seem to always terminate, but not always the dtmf is captured in a variable. 2. The read app seems to ignore the variable name parameter: calling it with 1 1 104.wav choice_181152 1 # doesn't put the digit in variable_choice_181152, but to dmtf_digit. Why is that? If it is coming from the sip provider as rfc 2833 dtmf, they are probably doing inband detection and failing at it. If you look at an rtp dump you can confirm this. If this is the case, there is nothing you can do on the FreeSWITCH side and the provider will have to fix it. But the call goes through the same SIP provider even when using the soft phone and there it works fine. The difference might be that then it is SIP to SIP within the same provider.. How do I do the RTP dump? Also I should have mentioned that DTMF is not captured only DURING the file is being played. It is always captured correctly when I wait until the playback is finished. Does this sound familiar? I thought this would be somet obvious misconfiguration on my side. Jan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
Hi, The send dtmf is working. thanks -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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: error: incompatible type for argument 3 of âswitch_loadable_module_unload_moduleâ mod_commands.c:869: error: too few arguments to function âswitch_loadable_module_unload_moduleâ mod_commands.c: In function âreload_functionâ: mod_commands.c:891: error: incompatible type for argument 3 of âswitch_loadable_module_unload_moduleâ mod_commands.c:891: error: too few arguments to function âswitch_loadable_module_unload_moduleâ cc1: warnings being treated as errors mod_commands.c: In function âtone_detect_session_functionâ: mod_commands.c:1089: warning: passing argument 6 of âswitch_ivr_tone_detect_sessionâ makes integer from pointer without a cast mod_commands.c:1089: error: too few arguments to function âswitch_ivr_tone_detect_sessionâ mod_commands.c: In function âuuid_send_dtmf_functionâ: mod_commands.c:2649: error: âstrâ undeclared (first use in this function) mod_commands.c:2649: error: (Each undeclared identifier is reported only once mod_commands.c:2649: error: for each function it appears in.) mod_commands.c:2649: error: expected â)â before string constant mod_commands.c:2649: error: too few arguments to function âstream-write_functionâ make[5]: *** [mod_commands.o] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_commands-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install+ +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 * *And another thing through default.xml barging is working fine and dtmf values also work fine But through api command i get the same error* api originate sofia/internal/1000%172.20.176.31 bridge(sofia/default/ [EMAIL PROTECTED]) Content-Type: api/response Content-Length: 41 +OK 81725d6b-c22f-4fb9-9e68-3e97eb1a3d4e api originate sofia/internal/1002%172.20.176.31eavesdrop(81725d6b-c22f-4fb9-9e68-3e97eb1a3d4e) Content-Type: api/response Content-Length: 41 +OK 1d0edac2-b548-4b02-80c3-46d7a1175845 api uuid_send_dtmf 1d0edac2-b548-4b02-80c3-46d7a1175845 2 to Content-Type: api/response Content-Length: 14 api uuid_send_dtmf 1d0edac2-b548-4b02-80c3-46d7a1175845 1 both Content-Type: api/response Content-Length: 14 -ERR no reply api uuid_send_dtmf 1d0edac2-b548-4b02-80c3-46d7a1175845 1 from Content-Type: api/response Content-Length: 14 -ERR no reply Correct me where i am wrong. Thanks for the reply. -- Warm Regards, N.Baskar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 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: error: incompatible type for argument 3 of âswitch_loadable_module_unload_moduleâ mod_commands.c:869: error: too few arguments to function âswitch_loadable_module_unload_moduleâ mod_commands.c: In function âreload_functionâ: mod_commands.c:891: error: incompatible type for argument 3 of âswitch_loadable_module_unload_moduleâ mod_commands.c:891: error: too few arguments to function âswitch_loadable_module_unload_moduleâ cc1: warnings being treated as errors mod_commands.c: In function âtone_detect_session_functionâ: mod_commands.c:1089: warning: passing argument 6 of âswitch_ivr_tone_detect_sessionâ makes integer from pointer without a cast mod_commands.c:1089: error: too few arguments to function âswitch_ivr_tone_detect_sessionâ mod_commands.c: In function âuuid_send_dtmf_functionâ: mod_commands.c:2649: error: âstrâ undeclared (first use in this function) mod_commands.c:2649: error: (Each undeclared identifier is reported only once mod_commands.c:2649: error: for each function it appears in.) mod_commands.c:2649: error: expected â)â before string constant mod_commands.c:2649: error: too few arguments to function âstream- write_functionâ make[5]: *** [mod_commands.o] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_commands-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install+ +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 8:30 AM, Gopala krishnan wrote: 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? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
Hi, I am using the event socket in freeswitch with audiocodes, and the client as a softphone. -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
I dont understand, can you please brief me? -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
Is there any dtmf setting that needs to be changed in the eyebeam phone? -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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, ___ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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}/${destination_number}/
action application=db data=insert/last_dial/global/$
{uuid}/
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Re: [Freeswitch-users] DTMF
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 that parameter Thanks in advance -- Warm Regards, N.Baskar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 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 -- -- Birgit Arkesteijn, [EMAIL PROTECTED], -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- URL: http://www.westhawk.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 -ERR no reply Then i passed all the values in the barging api uuid_send_dtmf baf82956-111d-4cd8-9568-47010ac8bd20 *2130** *output:* Content-Type: api/response Content-Length: 14 -ERR no reply Here i get only the beep sound for each values and error message as a output Thanks for the reply and correct me where i am work Thanks in advance. -- Warm Regards, N.Baskar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 i pass these through event socket it does not work why? if i pass the DTMF digits in api command it receive a beep sound but the process is not done. *i tried i n conference the output is :* api conference 3001 dtmf 15 # Content-Type: api/response Content-Length: 16 OK sent # to 15 15 is member id # is to hangup but it does not get hangup through api command * In barging also same problem * api uuid_send_dtmf f7666a65-5fc4-4199-bb6c-95a5e22d4515 3210* Content-Type: api/response Content-Length: 14 -ERR no reply I get the beep sound and there is no process done what should be done please guide me. Thanks in advance. -- Warm Regards, N.Baskar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF
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 manually done in softphone it work . when i press the # button it hangup and * for mute etc. it works fine but when i pass these through event socket it does not work why? if i pass the DTMF digits in api command it receive a beep sound but the process is not done. i tried i n conference the output is : api conference 3001 dtmf 15 # Content-Type: api/response Content-Length: 16 OK sent # to 15 15 is member id # is to hangup but it does not get hangup through api command In barging also same problem api uuid_send_dtmf f7666a65-5fc4-4199-bb6c-95a5e22d4515 3210* Content-Type: api/response Content-Length: 14 -ERR no reply I get the beep sound and there is no process done what should be done please guide me. Thanks in advance. -- Warm Regards, N.Baskar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF
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 uuid dtmf_data i did not know what is dtfm_data if dtmf_data is these DTMF signals values api uuid_send_dtmf b3df12bf-35c6-4a02-abaa-770c0d7bf358 *2* *output:* * Content-Type: api/response Content-Length: 14 -ERR no reply* I want to know what is dtmf_data? Any one correct me to solve my problem. -- Warm Regards, N.Baskar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF Star Event Inconsistent
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. Now, when i get a call on the soft phone, i press a sequence of DTMF digits. The sequence of DTMF digits is intended to be read by the Java code via the socket interface. Most things run pretty smoothly: i can place calls, i can send DTMFs, i can receive them on the other hand. The inconsistent behavior i'm seeing is following. For DTMF-0 to DTMF-9, and DTMF-#, i receive two events via the socket interface. The first one is in CS_EXCHANGE_MEDIA state and the second is in CS_EXECUTE state. Yet for DTMF-* i receive inconsistent number of events: sometimes only one single event in state CS_EXCHANGE_MEDIA sometimes two events as in the case of other DTMF digits. It seems there is a pattern in this inconsistency. The odd DTMF-* (first, third, fifth, etc.) generate only one CS_EXCHANGE_MEDIA event while the even (second, fourth, sixth, etc.) generate both events. Can somebody help me understand what's going on? Thanks, Klaus. -- Feel free - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: http://www.gmx.net/de/go/promail ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Star Event Inconsistent
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 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. Now, when i get a call on the soft phone, i press a sequence of DTMF digits. The sequence of DTMF digits is intended to be read by the Java code via the socket interface. Most things run pretty smoothly: i can place calls, i can send DTMFs, i can receive them on the other hand. The inconsistent behavior i'm seeing is following. For DTMF-0 to DTMF-9, and DTMF-#, i receive two events via the socket interface. The first one is in CS_EXCHANGE_MEDIA state and the second is in CS_EXECUTE state. Yet for DTMF-* i receive inconsistent number of events: sometimes only one single event in state CS_EXCHANGE_MEDIA sometimes two events as in the case of other DTMF digits. It seems there is a pattern in this inconsistency. The odd DTMF-* (first, third, fifth, etc.) generate only one CS_EXCHANGE_MEDIA event while the even (second, fourth, sixth, etc.) generate both events. Can somebody help me understand what's going on? Thanks, Klaus. -- Feel free - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: http://www.gmx.net/de/go/promail ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Star Event Inconsistent
I do indeed look for the Event-Name attribute. But since for a single DTMF digit two events are received from Freeswitch (with Event-Name: DTMF) , i need to differentiate them somehow such that one is processed and the other ignored. The differentiation pattern i found is the channel state (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] DTMF Star Event Inconsistent 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 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. Now, when i get a call on the soft phone, i press a sequence of DTMF digits. The sequence of DTMF digits is intended to be read by the Java code via the socket interface. Most things run pretty smoothly: i can place calls, i can send DTMFs, i can receive them on the other hand. The inconsistent behavior i'm seeing is following. For DTMF-0 to DTMF-9, and DTMF-#, i receive two events via the socket interface. The first one is in CS_EXCHANGE_MEDIA state and the second is in CS_EXECUTE state. Yet for DTMF-* i receive inconsistent number of events: sometimes only one single event in state CS_EXCHANGE_MEDIA sometimes two events as in the case of other DTMF digits. It seems there is a pattern in this inconsistency. The odd DTMF-* (first, third, fifth, etc.) generate only one CS_EXCHANGE_MEDIA event while the even (second, fourth, sixth, etc.) generate both events. Can somebody help me understand what's going on? Thanks, Klaus. -- Feel free - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: http://www.gmx.net/de/go/promail ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 at 11:36 AM, Klaus Teller [EMAIL PROTECTED] wrote: I do indeed look for the Event-Name attribute. But since for a single DTMF digit two events are received from Freeswitch (with Event-Name: DTMF) , i need to differentiate them somehow such that one is processed and the other ignored. The differentiation pattern i found is the channel state (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] DTMF Star Event Inconsistent 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 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. Now, when i get a call on the soft phone, i press a sequence of DTMF digits. The sequence of DTMF digits is intended to be read by the Java code via the socket interface. Most things run pretty smoothly: i can place calls, i can send DTMFs, i can receive them on the other hand. The inconsistent behavior i'm seeing is following. For DTMF-0 to DTMF-9, and DTMF-#, i receive two events via the socket interface. The first one is in CS_EXCHANGE_MEDIA state and the second is in CS_EXECUTE state. Yet for DTMF-* i receive inconsistent number of events: sometimes only one single event in state CS_EXCHANGE_MEDIA sometimes two events as in the case of other DTMF digits. It seems there is a pattern in this inconsistency. The odd DTMF-* (first, third, fifth, etc.) generate only one CS_EXCHANGE_MEDIA event while the even (second, fourth, sixth, etc.) generate both events. Can somebody help me understand what's going on? Thanks, Klaus. -- Feel free - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: http://www.gmx.net/de/go/promail ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Star Event Inconsistent
FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A00%3A47 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A00%3A47%20GMT Event-Date-timestamp: 1225144847932433 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1003%40192.168.50.94 Unique-ID: b35d1110-a472-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: G722 Channel-Read-Codec-Rate: 16000 Channel-Write-Codec-Name: G722 Channel-Write-Codec-Rate: 16000 DTMF-Digit: 8 DTMF-Duration: 2080 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A00%3A48 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A00%3A48%20GMT Event-Date-timestamp: 1225144848192750 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Original-Nachricht Datum: Mon, 27 Oct 2008 12:16:34 -0500 Von: Anthony Minessale [EMAIL PROTECTED] An: freeswitch-users@lists.freeswitch.org 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 at 11:36 AM, Klaus Teller [EMAIL PROTECTED] wrote: I do indeed look for the Event-Name attribute. But since for a single DTMF digit two events are received from Freeswitch (with Event-Name: DTMF) , i need to differentiate them somehow such that one is processed and the other ignored. The differentiation pattern i found is the channel state (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] DTMF Star Event Inconsistent 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 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. Now, when i get a call on the soft phone, i press a sequence of DTMF digits. The sequence of DTMF digits is intended to be read by the Java code via the socket interface. Most things run pretty smoothly: i can place calls, i can send DTMFs, i can receive them on the other hand. The inconsistent behavior i'm seeing is following. For DTMF-0 to DTMF-9, and DTMF-#, i receive two events via the socket interface. The first one is in CS_EXCHANGE_MEDIA state and the second is in CS_EXECUTE state. Yet for DTMF-* i receive inconsistent number of events: sometimes only one single event in state CS_EXCHANGE_MEDIA sometimes two events as in the case of other DTMF digits. It seems there is a pattern in this inconsistency. The odd DTMF-* (first, third, fifth, etc.) generate only one CS_EXCHANGE_MEDIA event while the even (second, fourth, sixth, etc.) generate both events. Can somebody help me understand what's going on? Thanks, Klaus. -- Feel free - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: http://www.gmx.net/de/go/promail ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 -- Pt
Re: [Freeswitch-users] DTMF Star Event Inconsistent
Interesting, thanks for pointing that. I would have thought that all events related to a call would have the same Unique-ID. Now I'm even more confused! regards, Klaus Original-Nachricht Datum: Mon, 27 Oct 2008 14:22:31 -0400 Von: Anthony Knight [EMAIL PROTECTED] 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 and Unique-ID: 34adac7a-a473-11dd-8207-2b46fcff01af You should only be looking for events on only one unique-id. If each ID is a leg of the call (Not sure about this) It also looks like each leg has a different sampling rate ie PCMU and G722. Hope this is helpful Tony Knight On Mon, Oct 27, 2008 at 2:08 PM, Klaus Teller [EMAIL PROTECTED] wrote: Thanks. I am not bridging any call. Calls are originated via the socket interface to the extension 1003. And for the same call, all digits except star will produce two events while star will produce one event sometimes and two events some other times in the same call. Here are for instance events i got in one single call, pressing 5** (five, star, star). You see that 5 produced two events, the first star produced one event, and the third star produced two events. Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: 5 DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A25 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A25%20GMT Event-Date-timestamp: 1225145065750884 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1003%40192.168.50.94 Unique-ID: 34adac7a-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: G722 Channel-Read-Codec-Rate: 16000 Channel-Write-Codec-Name: G722 Channel-Write-Codec-Rate: 16000 DTMF-Digit: 5 DTMF-Duration: 2080 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A26 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A26%20GMT Event-Date-timestamp: 1225145066008156 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: * DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A27 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A27%20GMT Event-Date-timestamp: 1225145067315160 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: * DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A28 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A28%20GMT Event-Date-timestamp: 1225145068213242 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1003
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 code. can you update to trunk before testing anymore? On Mon, Oct 27, 2008 at 1:32 PM, Klaus Teller [EMAIL PROTECTED] wrote: Interesting, thanks for pointing that. I would have thought that all events related to a call would have the same Unique-ID. Now I'm even more confused! regards, Klaus Original-Nachricht Datum: Mon, 27 Oct 2008 14:22:31 -0400 Von: Anthony Knight [EMAIL PROTECTED] 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 and Unique-ID: 34adac7a-a473-11dd-8207-2b46fcff01af You should only be looking for events on only one unique-id. If each ID is a leg of the call (Not sure about this) It also looks like each leg has a different sampling rate ie PCMU and G722. Hope this is helpful Tony Knight On Mon, Oct 27, 2008 at 2:08 PM, Klaus Teller [EMAIL PROTECTED] wrote: Thanks. I am not bridging any call. Calls are originated via the socket interface to the extension 1003. And for the same call, all digits except star will produce two events while star will produce one event sometimes and two events some other times in the same call. Here are for instance events i got in one single call, pressing 5** (five, star, star). You see that 5 produced two events, the first star produced one event, and the third star produced two events. Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: 5 DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A25 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A25%20GMT Event-Date-timestamp: 1225145065750884 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1003%40192.168.50.94 Unique-ID: 34adac7a-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: G722 Channel-Read-Codec-Rate: 16000 Channel-Write-Codec-Name: G722 Channel-Write-Codec-Rate: 16000 DTMF-Digit: 5 DTMF-Duration: 2080 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A26 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A26%20GMT Event-Date-timestamp: 1225145066008156 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: * DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A27 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A27%20GMT Event-Date-timestamp: 1225145067315160 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: * DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1
Re: [Freeswitch-users] DTMF Star Event Inconsistent
: 59c475e6-a484-11dd-93a2-e949eaab0e91 Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/internal/1003%40192.168.50.94 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1225152423965777 Caller-Channel-Created-Time: 1225152423965777 Caller-Channel-Answered-Time: 1225152426086251 Caller-Channel-Progress-Time: 1225152424066002 Caller-Channel-Progress-Media-Time: 1225152424066002 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false DTMF-Digit: * DTMF-Duration: 2080 Event-Name: DTMF Core-UUID: dab20b9c-a483-11dd-93a2-e949eaab0e91 FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 192.168.50.94 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2020%3A07%3A10 Event-Date-GMT: Tue,%2028%20Oct%202008%2000%3A07%3A10%20GMT Event-Date-timestamp: 1225152430335702 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 358 Original-Nachricht Datum: Mon, 27 Oct 2008 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 code. can you update to trunk before testing anymore? On Mon, Oct 27, 2008 at 1:32 PM, Klaus Teller [EMAIL PROTECTED] wrote: Interesting, thanks for pointing that. I would have thought that all events related to a call would have the same Unique-ID. Now I'm even more confused! regards, Klaus Original-Nachricht Datum: Mon, 27 Oct 2008 14:22:31 -0400 Von: Anthony Knight [EMAIL PROTECTED] 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 and Unique-ID: 34adac7a-a473-11dd-8207-2b46fcff01af You should only be looking for events on only one unique-id. If each ID is a leg of the call (Not sure about this) It also looks like each leg has a different sampling rate ie PCMU and G722. Hope this is helpful Tony Knight On Mon, Oct 27, 2008 at 2:08 PM, Klaus Teller [EMAIL PROTECTED] wrote: Thanks. I am not bridging any call. Calls are originated via the socket interface to the extension 1003. And for the same call, all digits except star will produce two events while star will produce one event sometimes and two events some other times in the same call. Here are for instance events i got in one single call, pressing 5** (five, star, star). You see that 5 produced two events, the first star produced one event, and the third star produced two events. Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003%40192.168.50.56%3A50435%3Brinstance%3D65055bcc835b9844 Unique-ID: 34b83622-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 DTMF-Digit: 5 DTMF-Duration: 2000 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A25 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A25%20GMT Event-Date-timestamp: 1225145065750884 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1003%40192.168.50.94 Unique-ID: 34adac7a-a473-11dd-8207-2b46fcff01af Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: G722 Channel-Read-Codec-Rate: 16000 Channel-Write-Codec-Name: G722 Channel-Write-Codec-Rate: 16000 DTMF-Digit: 5 DTMF-Duration: 2080 Event-Name: DTMF Core-UUID: 1abe8d52-a44b-11dd-8207-2b46fcff01af FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 127.0.0.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-10-27%2018%3A04%3A26 Event-Date-GMT: Mon,%2027%20Oct%202008%2022%3A04%3A26%20GMT Event-Date-timestamp: 1225145066008156 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_dequeue_dtmf Event-Calling-Line-Number: 357 Channel-State: CS_EXCHANGE_MEDIA Channel-State-Number: 5 Channel-Name: sofia/internal/1003
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 the socket interface to the extension 1003 and parked. Or does parking generate a second channel? I'm using Xlite to listen on 1003 and for sending DTMF digits on the parked channel. The wireshark trace also shows one single call going from my computer to the Freeswitch box located on another computer. And in this trace, events are not duplicated. I have updated and here is the new log information for 5** (DTMF-5, DTMF-*, DTMF-*) Klaus. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Star Event Inconsistent
I have the default Freeswitch settings. That is, Freeswitch as checked out from trunk (with event_socket.conf.xml changed to allow remote connections). I would thus think that this is RFC 2833. Also Wireshark is showing the DTMF gigits being sent as telephone-event which i guess sugesst 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 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 Xlite to listen on 1003 and for sending DTMF digits on the parked channel. The wireshark trace also shows one single call going from my computer to the Freeswitch box located on another computer. And in this trace, events are not duplicated. I have updated and here is the new log information for 5** (DTMF-5, DTMF-*, DTMF-*) Klaus. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Star Event Inconsistent
you clearly have 2 channels here. sofia/internal/[EMAIL PROTECTED] sofia/internal/[EMAIL PROTECTED]:50436;rinstance=15d12d876df10b6e if 1003 is a local user you should be originating with and @ try the % instead sofia/internal/1003%192.168.50.94 otherwise you are making a looped call over sip back to your own box. if you are using the default config you can also do user/1003 On Mon, Oct 27, 2008 at 3:32 PM, Klaus Teller [EMAIL PROTECTED] wrote: I have the default Freeswitch settings. That is, Freeswitch as checked out from trunk (with event_socket.conf.xml changed to allow remote connections). I would thus think that this is RFC 2833. Also Wireshark is showing the DTMF gigits being sent as telephone-event which i guess sugesst 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 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 Xlite to listen on 1003 and for sending DTMF digits on the parked channel. The wireshark trace also shows one single call going from my computer to the Freeswitch box located on another computer. And in this trace, events are not duplicated. I have updated and here is the new log information for 5** (DTMF-5, DTMF-*, DTMF-*) Klaus. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Reading and Playing
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]); s.hangup(); I can play the file 1.wav without problem but the send_dtmf is simply being ignored. I used wireshrack to check if maybe the outbound DTMF was sent and not played by my softphone. But this is not the case. streamFile() blocks until sound file ends or a DTMF tone is received, as detailed on the wiki: http://wiki.freeswitch.org/wiki/Session_streamFile I suspect you want some background music? I'm still trying to get my head around which programming features to use in which circumstances, something I've not found any clear high level guide on yet. begin:vcard fn:James Green n:Green;James org:StealthNET Ltd;Technical adr:;;Beacon Innovation Centre;Gorleston;Norfolk;NR31 7RA;GB email;internet:[EMAIL PROTECTED] tel;work:01493 66 00 32 url:http://www.stealthnet.net version:2.1 end:vcard ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
