Re: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-27 Thread Drew Ozier
Did a few TCP dumps, and before, I was seeing explicit DTMF packets coming
out of my FreeSWITCH machine. Removing the only things I could find that had
to do with 2833 in my internal and external sip_profiles (namely
rfc2833-pt), I find that I don't see any mention of switch_rtp.c sending any
DTMF packets, nor do I see any in the TCP dump.

This still hasn't solved my problem. I tested again, and with 'start_dtmf',
I get an audible hiccup in the audio when DTMF occurs. When I don't use
'start_dtmf', the hiccup goes away.

Is there anything else that I should be turning off on FreeSWITCH to avoid
this hiccup?

-Drew

On Wed, May 27, 2009 at 12:11 PM, Drew Ozier  wrote:

> The AudioCodes machines are always a bit of a nuisance to get acting the
> way I want, but I think I have them sending all DTMF in-band. How do I
> disable 2833 in my sofia profile?
>
> -Drew
>
>
> On Wed, May 27, 2009 at 11:52 AM, Brian West  wrote:
>
>> I highly recommend you DO NOT use inband AT ALL if possible.   out of band
>> is king.
>> /b
>>
>> On May 27, 2009, at 10:49 AM, Milena wrote:
>>
>> Are the configurations of the audiocodes set to send only in-audio dtmf?
>> make sure everything is sending only in-audio
>>
>>
>>   Brian West
>> br...@freeswitch.org
>>
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>>
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Re: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-27 Thread Drew Ozier
The AudioCodes machines are always a bit of a nuisance to get acting the way
I want, but I think I have them sending all DTMF in-band. How do I disable
2833 in my sofia profile?

-Drew


On Wed, May 27, 2009 at 11:52 AM, Brian West  wrote:

> I highly recommend you DO NOT use inband AT ALL if possible.   out of band
> is king.
> /b
>
> On May 27, 2009, at 10:49 AM, Milena wrote:
>
> Are the configurations of the audiocodes set to send only in-audio dtmf?
> make sure everything is sending only in-audio
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-27 Thread Drew Ozier
I'm sorry, far side meaning what exactly? The call flow is as follows:
TDM T1 -> AudioCodes Mediant 1000 -> FreeSWITCH -> AudioCodes Mediant 1000
-> TDM T1
FreeSWITCH gets an inbound SIP call and makes an outbound SIP call back to
the Mediant 1000, bridging the two calls. The Mediant 1000 sends the call
back out the T1 and that goes to my customer's IVR. I have no control over
anything after it leaves my T1.
Is there some other piece of information I'm missing? Do you want my
sip_profiles?

-Drew

On Wed, May 27, 2009 at 11:30 AM, Brian West  wrote:

> This is what starts the inband detector.. what is the far side?
>
> /b
>
>
> On May 27, 2009, at 10:25 AM, Drew Ozier wrote:
>
>  
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-27 Thread Drew Ozier
Hey Brian,

It is a bridged call. Here's the majority of my dialplan:



















On Wed, May 27, 2009 at 11:12 AM, Brian West  wrote:

> You should try to always use out of band... what is the call path because
> it looks like a bridged call and the far end gets the inband and the 2833
> /b
>
> On May 27, 2009, at 10:02 AM, Drew Ozier wrote:
>
> To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an
> AudioCodes Mediant 1000 (SIP Gateway) which goes to a FreeSWITCH machine. I
> receive calls coming in off the T1 which goes through my Mediant 1000 which
> goes to my FreeSWITCH machine which makes an outbound call back through the
> Mediant 1000 to an IVR (bridging the inbound and outbound calls). I have
> (supposedly) configured the AudioCodes Mediant 1000 to pass the DTMF
> in-band, because I want to minimize the amount of modifications I make to
> the audio stream going through my system. I also want to be able to log when
> DTMF events occur for analysis purposes. If I do not have start_dtmf, then
> FreeSWITCH does not show me any DTMF in the logs, but when I have it on, it
> will.
>
> Here's a DTMF event in my FreeSWITCH logs:
> (I changed the debug message to a notice and added the callId and the nano
> time of the event)
> 2009-05-26 16:02:05 [NOTICE] switch_ivr_async.c:996 inband_dtmf_callback()
> 348323 61182225861200042...@10.1.10.10 DTMF DETECTED: 5
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1322 do_2833() Send start packet
> for [5] ts=1217893935 dur=160/160/2000 seq=55199
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=320/320/2000 seq=55200
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=480/480/2000 seq=55201
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=640/640/2000 seq=55202
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=800/800/2000 seq=55203
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=960/960/2000 seq=55204
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=1120/1120/2000 seq=55205
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=1280/1280/2000 seq=55206
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=1440/1440/2000 seq=55207
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=1600/1600/2000 seq=55208
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=1760/1760/2000 seq=55209
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
> for [5] ts=1217893935 dur=1920/1920/2000 seq=55210
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for
> [5] ts=1217893935 dur=2080/2080/2000 seq=55211
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for
> [5] ts=1217893935 dur=2080/2080/2000 seq=55212
> 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for
> [5] ts=1217893935 dur=2080/2080/2000 seq=55213
>
> Does the fact that switch_rtp is sending packets mean that FreeSWITCH is
> sending out-of-band DTMF as well as recognizing the in-band?
>
> Thanks for your quick replies, hopefully this will make my issue clearer.
>
> -Drew
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-27 Thread Drew Ozier
To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an
AudioCodes Mediant 1000 (SIP Gateway) which goes to a FreeSWITCH machine. I
receive calls coming in off the T1 which goes through my Mediant 1000 which
goes to my FreeSWITCH machine which makes an outbound call back through the
Mediant 1000 to an IVR (bridging the inbound and outbound calls). I have
(supposedly) configured the AudioCodes Mediant 1000 to pass the DTMF
in-band, because I want to minimize the amount of modifications I make to
the audio stream going through my system. I also want to be able to log when
DTMF events occur for analysis purposes. If I do not have start_dtmf, then
FreeSWITCH does not show me any DTMF in the logs, but when I have it on, it
will.

Here's a DTMF event in my FreeSWITCH logs:
(I changed the debug message to a notice and added the callId and the nano
time of the event)
2009-05-26 16:02:05 [NOTICE] switch_ivr_async.c:996 inband_dtmf_callback()
348323 61182225861200042...@10.1.10.10 DTMF DETECTED: 5
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1322 do_2833() Send start packet
for [5] ts=1217893935 dur=160/160/2000 seq=55199
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=320/320/2000 seq=55200
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=480/480/2000 seq=55201
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=640/640/2000 seq=55202
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=800/800/2000 seq=55203
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=960/960/2000 seq=55204
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=1120/1120/2000 seq=55205
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=1280/1280/2000 seq=55206
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=1440/1440/2000 seq=55207
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=1600/1600/2000 seq=55208
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=1760/1760/2000 seq=55209
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet
for [5] ts=1217893935 dur=1920/1920/2000 seq=55210
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for
[5] ts=1217893935 dur=2080/2080/2000 seq=55211
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for
[5] ts=1217893935 dur=2080/2080/2000 seq=55212
2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for
[5] ts=1217893935 dur=2080/2080/2000 seq=55213

Does the fact that switch_rtp is sending packets mean that FreeSWITCH is
sending out-of-band DTMF as well as recognizing the in-band?

Thanks for your quick replies, hopefully this will make my issue clearer.

-Drew

On Wed, May 27, 2009 at 10:22 AM, Milena  wrote:

> Check the freeswitch log when the dtmf are being received, most likely your
> device is sending them both as Dave says, maybe the telephone you're dialing
> with sends it both inband and rtp. What does the log says?
>
> 2009/5/26 Drew Ozier 
>
>> I've got a configuration where I receive inbound calls and dial out to a
>> pre-determined 800-number based on the DNIS of the call. I set > application="start_dtmf"/> and have everything set up so that DTMF only
>> comes to me via inband. When I'm providing DTMF data to the IVR, it will
>> recognize a single keypress as a double-tap. My FreeSWITCH logs only contain
>> one DTMF entry, but when I listen to the receiving end of the call, I can
>> hear a hiccup in the DTMF tone that is getting played. When I do not use
>> 'start_dtmf', this problem goes away. I need inband DTMF detection, but I
>> can't have it messing up the audio stream. Any thoughts?
>>
>> -Drew Ozier
>>
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[Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH

2009-05-26 Thread Drew Ozier
I've got a configuration where I receive inbound calls and dial out to a
pre-determined 800-number based on the DNIS of the call. I set  and have everything set up so that DTMF only
comes to me via inband. When I'm providing DTMF data to the IVR, it will
recognize a single keypress as a double-tap. My FreeSWITCH logs only contain
one DTMF entry, but when I listen to the receiving end of the call, I can
hear a hiccup in the DTMF tone that is getting played. When I do not use
'start_dtmf', this problem goes away. I need inband DTMF detection, but I
can't have it messing up the audio stream. Any thoughts?

-Drew Ozier
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[Freeswitch-users] Problem listening to PCMU and ul recordings

2009-04-07 Thread Drew Ozier
Hi,

I've been using the record_session feature and wish to use PCMU or ul audio
format, but when I try to play back the audio in either format, it sounds
high-pitch and fast as if it is playing back at 2x speed. I looked at the
waveform recorded in PCMU and ul versus what it looks like when I record as
wav, and it seems like it is only recording every-other sample (which would
explain the pitch and speed).

My vars.xml is set to PCMU as the global codec pref and the outbound codec
pref.

I am recording in stereo (one channel per leg of the call), but I'm not
messing with any other recording parameters.

Incedentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio
file, because I'm getting calls off a T1 (actually off of an AudioCodes
Mediant 1000 that is converting the T1 to SIP for me), and I'd like to
record precicely what is coming off the wire.

I'd be happy to send any configuration files, I'm just currently at a loss
for how to proceed.

Thanks,

Drew Ozier
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[Freeswitch-users] Problem listening to PCMU and ul recordings

2009-04-07 Thread Drew Ozier
Hi,

I've been using the record_session feature and wish to use PCMU or ul audio
format, but when I try to play back the audio in either format, it sounds
high-pitch and fast as if it is playing back at 2x speed. I looked at the
waveform recorded in PCMU and ul versus what it looks like when I record as
wav, and it seems like it is only recording every-other sample (which would
explain the pitch and speed).

My vars.xml is set to PCMU as the global codec pref and the outbound codec
pref.

I am recording in stereo (one channel per leg of the call), but I'm not
messing with any other recording parameters.

Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio
file, because I'm getting calls off a T1 (actually off of an AudioCodes
Mediant 1000 that is converting the T1 to SIP for me), and I'd like to
record precisely what is coming off the wire.

I'd be happy to send any configuration files, I'm just currently at a loss
for how to proceed.

Thanks,

Drew Ozier
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[Freeswitch-users] Problem listening to PCMU and ul recordings

2009-04-07 Thread Drew Ozier
Hi,

I've been using the record_session feature and wish to use PCMU or ul audio
format, but when I try to play back the audio in either format, it sounds
high-pitch and fast as if it is playing back at 2x speed. I looked at the
waveform recorded in PCMU and ul versus what it looks like when I record as
wav, and it seems like it is only recording every-other sample (which would
explain the pitch and speed).

My vars.xml is set to PCMU as the global codec pref and the outbound codec
pref.

I am recording in stereo (one channel per leg of the call), but I'm not
messing with any other recording parameters.

Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio
file, because I'm getting calls off a T1 (actually off of an AudioCodes
Mediant 1000 that is converting the T1 to SIP for me), and I'd like to
record precisely what is coming off the wire.

I'd be happy to send any configuration files, I'm just currently at a loss
for how to proceed.

Thanks,

Drew Ozier
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