Re: [asterisk-users] DTMF rfc2833 missed when transfering to another server

[Israel Gottlieb]

well looks likes we solved it
the rtpkeepalive was set to 5 seconds on the trunk and every time asterisk
sends a rtpkeepalive a cn packet is sent
the same time a cn packet is sent asterisk loses the dtmf it was sent


On Wed, Dec 16, 2020 at 7:43 PM Israel Gottlieb  wrote:

> Hi all
> i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and
> then calls a second server (leg B) server B is a freeswitch server
>
> the servers are configured all thru with rfc2833 for dtmf
> the caller enters a number a long 15 digit number like a credit card
> number or even a phone number and in alot of cases server B always
> doesn't get part of the digits from server A
>
> running a trace on server (A) i checked the trace of leg A and of leg B on
> the same server (A) and i see that the from the provider to asterisk has
> all digits correct but leg b going out the same server has a missed digit
>
> so either asterisk isnt getting all digits from the provider for some
> reason or it fails to regenerate the dtmf when sending to server b
> another think i noticed is asterisk generating a rtp  (cn) packet to leg
> every time it misses
>
> any idea how i can check what asterisk is seeing if its just sending the
> rtp without transcoding ?
> does anyone have a idea of what might be the problem
>
> using chan_sip
> rfc2833compensate=yes
> relaxdtmf=yes
>
> thanks
>
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Re: [asterisk-users] DTMF not working on incoming calls

[Dovid Bender]

Have you done a wireshark capture and then seen if the DTMF is coming in
from your provider? What does the SDP show?



On Thu, Dec 5, 2019 at 12:17 AM Carlos Chavez  wrote:

> What is  the best way to debug DTMF on a PJSIP trunk?  I have cycled
> through all available options ('rfc4733','inband','info','auto','auto_info')
> but my IVR does not recognize any options from the remote end.  I have also
> tried changing codecs from g729 to alaw or ulaw with the same result.
> Outgoing calls do not seem to have this problem, just incoming.  This is
> with Asterisk 13.29.2 but the problem started with 13.21 before I decided
> to upgrade to the latest 13.x version.  Any pointers?
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Re: [asterisk-users] DTMF tones in MixMonitor recording

[Patrick Wakano]

I agree! I have my SBC and asterisk servers all configured with rfc2833, so
it should be ok! No need for auto mode!
Thanks again!
Cheers
Patrick

On Tue, 1 May 2018, 20:07 Joshua Colp,  wrote:

> On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote:
> > Thanks very much for the reply Joshua!
> > So I guess that setting dtmfmode=auto would be the safest choice in order
> > to strip out the DTMFs from the recording, right?
> > Cheers!
>
> It should work. Personally I prefer explicit configuring instead of having
> things just try to figure out what is in use.
>
> --
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> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] DTMF tones in MixMonitor recording

[Joshua Colp]

On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote:
> Thanks very much for the reply Joshua!
> So I guess that setting dtmfmode=auto would be the safest choice in order
> to strip out the DTMFs from the recording, right?
> Cheers!

It should work. Personally I prefer explicit configuring instead of having 
things just try to figure out what is in use.

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Re: [asterisk-users] DTMF tones in MixMonitor recording

[Patrick Wakano]

Thanks very much for the reply Joshua!
So I guess that setting dtmfmode=auto would be the safest choice in order
to strip out the DTMFs from the recording, right?
Cheers!
Patrick Wakano

On Tue, 1 May 2018, 19:36 Joshua Colp,  wrote:

> On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote:
> > Hello list,
> > Hope you are all doing fine!
> >
> > I have stumbled over some piece of dialplan code in which apparently they
> > were trying to avoid recording the DTMF tones in the wav file. It is
> really
> > messy and I am not sure if this really works. So after a bit of research
> I
> > found this comment (
> > https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which
> it is
> > said:
> >
> > *"Asterisk strips the DTMF from the audio stream when configured for
> > inband, so internal stuff can react to the DTMF and so the other side
> does
> > not hear the tone unless they are using inband (in which case it is
> > regenerated)"*
> > So my questions are, what are the cases in which Asterisk regenerates the
> > DTMFs? Does it cause the recording to have the tone as well, or is it
> only
> > transmitted to the other leg without being generated to the recording
> file?
> > Also, what if one or both legs are RFC2833? From my tests the RFC2833
> > events never show up in the recording, but I just want to confirm that
> this
> > is always true.
>
> If properly configured then Asterisk will always strip and regenerate the
> DTMF tone. You have to purposely misconfigure things to cause it to not get
> stripped. IE: DTMF is actually inband but you configure it for RFC2833.
> Since Asterisk wouldn't be listening to the audio stream, it would go right
> through and get recorded.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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>
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Re: [asterisk-users] DTMF tones in MixMonitor recording

[Joshua Colp]

On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote:
> Hello list,
> Hope you are all doing fine!
> 
> I have stumbled over some piece of dialplan code in which apparently they
> were trying to avoid recording the DTMF tones in the wav file. It is really
> messy and I am not sure if this really works. So after a bit of research I
> found this comment (
> https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is
> said:
> 
> *"Asterisk strips the DTMF from the audio stream when configured for
> inband, so internal stuff can react to the DTMF and so the other side does
> not hear the tone unless they are using inband (in which case it is
> regenerated)"*
> So my questions are, what are the cases in which Asterisk regenerates the
> DTMFs? Does it cause the recording to have the tone as well, or is it only
> transmitted to the other leg without being generated to the recording file?
> Also, what if one or both legs are RFC2833? From my tests the RFC2833
> events never show up in the recording, but I just want to confirm that this
> is always true.

If properly configured then Asterisk will always strip and regenerate the DTMF 
tone. You have to purposely misconfigure things to cause it to not get 
stripped. IE: DTMF is actually inband but you configure it for RFC2833. Since 
Asterisk wouldn't be listening to the audio stream, it would go right through 
and get recorded.

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] DTMF emulation with SIP INFO and direct media

[Jean Aunis]

Asterisk is in version 14.7.1. One end is a SIP Trunk to another 
Asterisk, the other end a home-made SIP phone. SIP INFO requests are 
coming from the other Asterisk.


Both endpoints use chan_sip with "dtmfmode" set to "info".

This is not strictly speaking a one-to-one setup since we're connecting 
to a SIP Trunk which then connects to another SIP phone, but I think it 
doesn't make much difference regarding SIP INFO handling.



Le 15/12/2017 à 12:12, Olivier a écrit :

Hello Jean,

1. Can you describe a bit further how both ends of the above call were 
both made of and configured ?

DTMF receiving is Asterisk/SIP channel but which version ?
Is the other end a SIP phone or a SIP trunk ?

2. Do you observe such behaviour in a one-to-one setup (one end emits, 
the other listen) or does the DTMF sending side also communicates with 
an other endpoint ?


Cheers

2017-12-13 12:22 GMT+01:00 Jean Aunis >:


Hello,

I think there is an issue when DTMF are handled with SIP INFO and
direct media is enabled.

When I receive a SIP INFO, the logs tell me that a "DTMF begin" is
generated, but no related "DTMF end" is generated, unless the call
is ended. Here is an excerpt of the logs :

*--- SIP INFO received **on **SIP/xxx-0004:*

[Dec 13 11:56:16] DTMF[18193][C-0005] channel.c: DTMF end '#'
received on SIP/xxx-0004, duration 257 ms
[Dec 13 11:56:16] DTMF[18193][C-0005] channel.c: DTMF begin
emulation of '#' with duration 257 queued on SIP/xxx-0004

*--- **SIP/xxx-0004 **is hanged up:*

[Dec 13 11:56:19] VERBOSE[18193][C-0005] bridge_channel.c:
Channel SIP/xxx-0004 left 'native_rtp' basic-bridge
<4a5905ac-29f8-41c5-9981-e9d0f4966c56>
[Dec 13 11:56:19] DTMF[18193][C-0005] bridge_channel.c: DTMF
end '#' simulated to bridge 4a5905ac-29f8-41c5-9981-e9d0f4966c56
because SIP/xxx-0004 left.  Duration 3012 ms.

Do you think it is a bug ? I would tend to say yes, but I'm not so
sure.

Regards

Jean Aunis


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Re: [asterisk-users] DTMF emulation with SIP INFO and direct media

[Olivier]

Hello Jean,

1. Can you describe a bit further how both ends of the above call were both
made of and configured ?
DTMF receiving is Asterisk/SIP channel but which version ?
Is the other end a SIP phone or a SIP trunk ?

2. Do you observe such behaviour in a one-to-one setup (one end emits, the
other listen) or does the DTMF sending side also communicates with an other
endpoint ?

Cheers

2017-12-13 12:22 GMT+01:00 Jean Aunis :

> Hello,
>
> I think there is an issue when DTMF are handled with SIP INFO and direct
> media is enabled.
>
> When I receive a SIP INFO, the logs tell me that a "DTMF begin" is
> generated, but no related "DTMF end" is generated, unless the call is
> ended. Here is an excerpt of the logs :
>
> *--- SIP INFO received **on **SIP/xxx-0004:*
>
> [Dec 13 11:56:16] DTMF[18193][C-0005] channel.c: DTMF end '#' received
> on SIP/xxx-0004, duration 257 ms
> [Dec 13 11:56:16] DTMF[18193][C-0005] channel.c: DTMF begin emulation
> of '#' with duration 257 queued on SIP/xxx-0004
>
> *--- **SIP/xxx-0004 **is hanged up:*
>
> [Dec 13 11:56:19] VERBOSE[18193][C-0005] bridge_channel.c: Channel
> SIP/xxx-0004 left 'native_rtp' basic-bridge <4a5905ac-29f8-41c5-9981-
> e9d0f4966c56>
> [Dec 13 11:56:19] DTMF[18193][C-0005] bridge_channel.c: DTMF end '#'
> simulated to bridge 4a5905ac-29f8-41c5-9981-e9d0f4966c56 because
> SIP/xxx-0004 left.  Duration 3012 ms.
>
> Do you think it is a bug ? I would tend to say yes, but I'm not so sure.
>
> Regards
>
> Jean Aunis
>
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Re: [asterisk-users] DTMF issues between Asterisk and Callmanager with Zoiper

[Joshua Colp]

Carlos Chavez wrote:

I had an old Asterisk installation die recently and we decided to
upgrade to Asterisk 13 to replace the old server. Everything seems to be
working with PJSIP but there is one issue. Asterisk talks to a
callmanager via a SIP trunk and send calls to PSTN (another country).
Most of the calls that go through there are for conferences. Desk phones
can enter the conferences without any issues but users with softphones
like Zoiper cannot. The conference systems either duplicate digits or
drop some. I have tried using inband, info and rfc4733 but the
softphones always have the same problem.

Anyone has any experience with softphone dtmf issues?


While I can't say I've had problems I can say you might want to try to 
narrow things down a bit. Do testing strictly to Asterisk first using 
something like Read and see if that is fine. If so then you've narrowed 
it down to the outgoing side, which may mean that the length of digits 
or something else produced by the softphone is not liked by it.


Cheers,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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Re: [asterisk-users] DTMF talkoff beep (still)

[Pete Mundy]

On 9/10/2015, at 5:16 AM, Jamie Rees  wrote:

> 
> 
> I understand this is DTMF talkoff
> 
> 
> My question is how do people running SIP phone systems mitigate against this?


My personal answer to this question has been to completely avoid the use of any 
ATAs at all. Since taking that approach I have never had to deal with the 
problem again.

Not sure how much practical use that is to you in your own situation, but it 
worked for me!

Pete

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Re: [asterisk-users] DTMF issue

[Jamie Rees]

Hi all, 

Just an update here

I added the relaxdtmf=yes in sip_custom.conf, given according to the
documentation the default option is no. This has made a bit of difference,
I'm getting less reports of it now although one particular person seems to
still be affected (they talk to a particular person who has a distinctive
voice)

All phones are Cisco SPA512G, are set to G711u/ulaw codec, DTMF process
AVT=yes and DTMF TX Method: Auto. I tried InBand, amongst others and that
did nothing. 

Where some DTMF bursts are lower than 80ms (which is the lowest Asterisk
expects), it's triggering emulation to bring it in line. I've read that
reducing the minimum DTMF tone length to 40ms can solve this issue, by
editing the #define AST_MIN_DTMF_DURATION variable. 

Does anyone concur? If so, where can I find said variable in the config
files? 

Thanks again,

Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: 08 July 2015 10:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

Indeed, thanks.
I'll let you know how it goes. 
Thanks,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 22:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

You probably have to reload asrerisk  after making the change. 

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 3:53 PM >>>
Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.

I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.

Thanks,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."

The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn&

Re: [asterisk-users] DTMF issue

[Jamie Rees]

Indeed, thanks.
I'll let you know how it goes. 
Thanks,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 22:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

You probably have to reload asrerisk  after making the change. 

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 3:53 PM >>>
Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.

I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.

Thanks,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."

The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I p

Re: [asterisk-users] DTMF issue

[Tom Peters]

You probably have to reload asrerisk  after making the change. 

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 3:53 PM >>>
Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.

I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.

Thanks,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF 
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."

The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:2

Re: [asterisk-users] DTMF issue

[Jamie Rees]

Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.

I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.

Thanks,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."

The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF

Re: [asterisk-users] DTMF issue

[Tom Peters]

In my humble opinion, adjusting this setting will (for you) do nothing, since 
you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF 
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in sip.conf."

The big question for you is going to be, does your system need to recognize 
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems doing 
that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-000

Re: [asterisk-users] DTMF issue

[Jamie Rees]

Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation
of '3' queued on SIP/209-00021cac

 

Can someone please provide any tips? 

 

Thanks,

Jamie 



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Re: [asterisk-users] DTMF issue

[Tom Peters]

It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile 
phones but it happens at random on many external calls. If this happens to you, 
especially on voice peaks (when the outside party said a particularly loud 
syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a broken 
DTMF tone and getting regenerated by your T1 or POTS card, or Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by using 
...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone from 
callers very well. They would dial 4 digits and in my logs, I'd see one or two, 
maybe three. The autoattendant would tell them they had dialed an invalid 
extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So 
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation
of '3' queued on SIP/209-00021cac

 

Can someone please provide any tips? 

 

Thanks,

Jamie 



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Re: [asterisk-users] DTMF issue

[Andres]

On 7/6/15 5:53 PM, Jamie Rees wrote:


Hello folks,

We have an issue with several Cisco SPA512G phones connected to an 
Asterisk platform where several users hear loud, random beeps during 
calls to external recipients. The noises are akin to button press 
tones, are very loud and a significant annoyance.


I've tried changing the DTMF tones on the phones (512G's running 
firmware 7.5.5) from In-Band to every other possibility, but this 
hasn't helped at all. The provider has suggested RFC2833 out-of-band, 
but the Cisco manuals do not clearly state which setting this is on 
the handsets.


I have enabled DTMF logging and spoken to the SIP provider, but they 
couldn't really help much. I presume the issue is local to our phone 
system but other than the logs below, have nothing to go on:


[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2' 
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin 
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2' 
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end 
accepted with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end 
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3' 
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin 
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' 
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end 
accepted with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' 
detected to have actual duration 78 on the wire, emulation will be 
triggered on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' 
has duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end 
emulation of '3' queued on SIP/209-00021cac


Can someone please provide any tips?

Yes, I have had this annoyance happen to me before.  It is very 
frustrating.   In order to rule out the SIP Provider, I suggest you 
record the call.  If the beep is not heard in the recording but only by 
the end user on the Cisco Phone, then its a phone issue.  The phone is 
confusing audio with the specific frequencies of DTMF. There is little 
you can do to fix this except for firmware upgrades (and I remember 
there were some that addressed this specific issue, at least on Cisco ATAs).


Thanks,

Jamie






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Re: [asterisk-users] DTMF issue

[Ryan, Travis]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: Monday, July 06, 2015 5:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue

Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk 
platform where several users hear loud, random beeps during calls to external 
recipients. The noises are akin to button press tones, are very loud and a 
significant annoyance.

I've tried changing the DTMF tones on the phones (512G's running firmware 
7.5.5) from In-Band to every other possibility, but this hasn't helped at all. 
The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not 
clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they couldn't 
really help much. I presume the issue is local to our phone system but other 
than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2' received 
on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin passthrough 
'2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2' received 
on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted with 
begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough 
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3' received 
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin passthrough 
'3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' received 
on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted with 
begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' detected 
to have actual duration 78 on the wire, emulation will be triggered on 
SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has 
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation of 
'3' queued on SIP/209-00021cac

Can someone please provide any tips?

Thanks,
Jamie


This doesn't help, but It DOES sound familiar. I've not seen this for a long 
time. If I can remember I'll write back. Just thought I'd let you know you're 
not crazy. :)
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Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

[Yaron Nachum]

Thank you Mathew,
We tested the feature flag workaround and it worked.

We opened a ticket - Asterisk-24459.

If you need any information please get back to us and we will do our best.

Thanks again,
Yaron.

On Mon, Oct 27, 2014 at 3:48 PM, Matthew Jordan  wrote:

>
>
> On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum 
> wrote:
>
>> Hello Mathew,
>> Thank you for the reply.
>>
>> I will open an issue and send debug information.
>>
>> Can you explain more about the workaround? A reference to the
>> documentation would be fine.
>>
>>
>>
> Sure - really, what you are running into is a difference in how Asterisk
> bridges channels:
>
> https://wiki.asterisk.org/wiki/display/AST/Bridges
>
> I suspect the reason DTMF is not decoded is because you are in a native
> bridge (local or remote). You can force a core two-party bridge by
> requiring that Asterisk decode the media and detect DTMF. Those
> requirements are done by setting the various 'feature' flags on whatever
> dialplan application is causing the channels to be bridged. For an example,
> see the 't' or 'T' options in Dial:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

[Matthew Jordan]

On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum 
wrote:

> Hello Mathew,
> Thank you for the reply.
>
> I will open an issue and send debug information.
>
> Can you explain more about the workaround? A reference to the
> documentation would be fine.
>
>
>
Sure - really, what you are running into is a difference in how Asterisk
bridges channels:

https://wiki.asterisk.org/wiki/display/AST/Bridges

I suspect the reason DTMF is not decoded is because you are in a native
bridge (local or remote). You can force a core two-party bridge by
requiring that Asterisk decode the media and detect DTMF. Those
requirements are done by setting the various 'feature' flags on whatever
dialplan application is causing the channels to be bridged. For an example,
see the 't' or 'T' options in Dial:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

[Yaron Nachum]

Hello Mathew,
Thank you for the reply.

I will open an issue and send debug information.

Can you explain more about the workaround? A reference to the documentation
would be fine.


Thanks again,
Yaron.

On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan  wrote:

>
>
> On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum 
> wrote:
>
>> Hello all,
>> We have recently upgraded some of our services to Asterisk 12 with PJSIP.
>> We have 2 issues related to DTMF:
>> 1. in the regular SIP channel we had DTMF auto mode, which adapted the
>> DTMF settings according to the incoming INVITE - RFC2833 or inband. The is
>> no such settings in PJSIP. Do you know is there is a plan to develop it?
>>
>
> No one that I'm aware of is currently working on that.
>
> As Asterisk is an open source project, if having the 'auto' feature added
> to the PJSIP stack is something you're interested in, you should consider
> writing a patch for the project [1].
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
>
>
>> 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk
>> does not transcode the DTMF signals, therefore DTMF is not working. It used
>> to work on release 11. This is really bad. Do you know of a solution to
>> this issue? Maybe some settings?
>>
>>
> That actually is a bug. You are most likely ending up in a native packet
> to packet bridge (or a native remote bridge), which does not decode the RTP
> stream. Hence, the inband DTMF or RFC 2833 DTMF is not being decoded and is
> being passed to the other side. Please do open an issue for that [2]. Make
> sure you provide a full DEBUG log, as that will illustrate what is actually
> occurring.
>
> Note that you can work around that issue by adding a feature flag to
> whatever application caused the bridging to occur.
>
> [2] https://issues.asterisk.org/jira
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

[Matthew Jordan]

On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum 
wrote:

> Hello all,
> We have recently upgraded some of our services to Asterisk 12 with PJSIP.
> We have 2 issues related to DTMF:
> 1. in the regular SIP channel we had DTMF auto mode, which adapted the
> DTMF settings according to the incoming INVITE - RFC2833 or inband. The is
> no such settings in PJSIP. Do you know is there is a plan to develop it?
>

No one that I'm aware of is currently working on that.

As Asterisk is an open source project, if having the 'auto' feature added
to the PJSIP stack is something you're interested in, you should consider
writing a patch for the project [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process


> 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk
> does not transcode the DTMF signals, therefore DTMF is not working. It used
> to work on release 11. This is really bad. Do you know of a solution to
> this issue? Maybe some settings?
>
>
That actually is a bug. You are most likely ending up in a native packet to
packet bridge (or a native remote bridge), which does not decode the RTP
stream. Hence, the inband DTMF or RFC 2833 DTMF is not being decoded and is
being passed to the other side. Please do open an issue for that [2]. Make
sure you provide a full DEBUG log, as that will illustrate what is actually
occurring.

Note that you can work around that issue by adding a feature flag to
whatever application caused the bridging to occur.

[2] https://issues.asterisk.org/jira

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] DTMF transmitting letter A

[Eric Wieling]

A is a valid DTMF "digit", chances are your PBX is detecting the digit wrong.   
If you have relaxdtmf enabled, disable it.   If that doesn't help, play with 
the audio gains.  Too loud or too soft can cause DTMF issues.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Tuesday, June 17, 2014 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DTMF transmitting letter A

Dear list,

maybe not really an Asterisk question, but... all my users dial in via 
PSTN (via SIP DIDs) and enter a target number via DTMF through my 
Asterisk 1.4. Out of about 150,000 calls per month I see on average 
about 1 call per month where an arbitrary caller enters the letter 'A' 
via DTMF. These callers use their mobile phones to dial in. I just 
reread the Wikipedia article on DTMF but I don't understand how someone 
can send an 'A'. Any clue?

Thank you!
Markus

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Re: [asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

Let me try with dtmfmode as auto...
On 28 May 2013 19:32, "Asghar Mohammad"  wrote:

> work around was block dtmf.
> set wrong type of dtmf in incoming trunk.
>
>
> On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> So any resolution for this?
>>
>> I suspect it could be related to RE INVITE
>>
>>
>> On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote:
>>
>>> i had this in past there was an ATA configured to send 9 at the end of
>>> dialing in my case.
>>>
>>>
>>> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
 Hi,

 I am receiving DTMF without any reason after call establishment.

 The log as follows, and I suspect something related to directmedia,
 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
 is making progress passing it to SIP/MAN-000a4b48
 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
 answered SIP/MAN-000a4b48
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
 SIP/MyTrunk-000a4b49, duration 0 ms
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
 '*' on SIP/MyTrunk-000a4b49
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
 SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
 SIP/MyTrunk-000a4b49, duration 0 ms
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
 '8' on SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
 SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
 SIP/MAN-000a4af0, duration 100 ms
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8'
 with duration 100 queued on SIP/MAN-000a4af0
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8'
 queued on SIP/MAN-000a4af0
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
 SIP/MAN-000a4b41, duration 100 ms
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1'
 with duration 100 queued on SIP/MAN-000a4b41
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1'
 queued on SIP/MAN-000a4b41
 [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
 (sip-trunk-inbound, 2127773456, 1) exited non-zero on
 'SIP/MyTrunk-000a4af3'
 [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing
 [h@trunk-outbound:1] NoOp("SIP/MAN-000a4b09", "16") in new stack
 [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
 (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'

 Is this some thing related to SIP RE-INVITE?

 Thanks.


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>>>
>>>
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>>
>>
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Re: [asterisk-users] DTMF recognized after call establishment

[Asghar Mohammad]

work around was block dtmf.
set wrong type of dtmf in incoming trunk.


On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> So any resolution for this?
>
> I suspect it could be related to RE INVITE
>
>
> On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote:
>
>> i had this in past there was an ATA configured to send 9 at the end of
>> dialing in my case.
>>
>>
>> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I am receiving DTMF without any reason after call establishment.
>>>
>>> The log as follows, and I suspect something related to directmedia,
>>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>>> is making progress passing it to SIP/MAN-000a4b48
>>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>>> answered SIP/MAN-000a4b48
>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
>>> SIP/MyTrunk-000a4b49, duration 0 ms
>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
>>> '*' on SIP/MyTrunk-000a4b49
>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
>>> SIP/MyTrunk-000a4b49
>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
>>> SIP/MyTrunk-000a4b49, duration 0 ms
>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
>>> '8' on SIP/MyTrunk-000a4b49
>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
>>> SIP/MyTrunk-000a4b49
>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
>>> SIP/MAN-000a4af0, duration 100 ms
>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
>>> duration 100 queued on SIP/MAN-000a4af0
>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued
>>> on SIP/MAN-000a4af0
>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
>>> SIP/MAN-000a4b41, duration 100 ms
>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
>>> duration 100 queued on SIP/MAN-000a4b41
>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued
>>> on SIP/MAN-000a4b41
>>> [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
>>> (sip-trunk-inbound, 2127773456, 1) exited non-zero on
>>> 'SIP/MyTrunk-000a4af3'
>>> [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
>>> NoOp("SIP/MAN-000a4b09", "16") in new stack
>>> [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
>>> (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'
>>>
>>> Is this some thing related to SIP RE-INVITE?
>>>
>>> Thanks.
>>>
>>>
>>> --
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>>
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Re: [asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

So any resolution for this?

I suspect it could be related to RE INVITE


On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote:

> i had this in past there was an ATA configured to send 9 at the end of
> dialing in my case.
>
>
> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi,
>>
>> I am receiving DTMF without any reason after call establishment.
>>
>> The log as follows, and I suspect something related to directmedia,
>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>> is making progress passing it to SIP/MAN-000a4b48
>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>> answered SIP/MAN-000a4b48
>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
>> SIP/MyTrunk-000a4b49, duration 0 ms
>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
>> '*' on SIP/MyTrunk-000a4b49
>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
>> SIP/MyTrunk-000a4b49
>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
>> SIP/MyTrunk-000a4b49, duration 0 ms
>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
>> '8' on SIP/MyTrunk-000a4b49
>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
>> SIP/MyTrunk-000a4b49
>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
>> SIP/MAN-000a4af0, duration 100 ms
>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
>> duration 100 queued on SIP/MAN-000a4af0
>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued
>> on SIP/MAN-000a4af0
>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
>> SIP/MAN-000a4b41, duration 100 ms
>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
>> duration 100 queued on SIP/MAN-000a4b41
>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued
>> on SIP/MAN-000a4b41
>> [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
>> (sip-trunk-inbound, 2127773456, 1) exited non-zero on
>> 'SIP/MyTrunk-000a4af3'
>> [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
>> NoOp("SIP/MAN-000a4b09", "16") in new stack
>> [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
>> (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'
>>
>> Is this some thing related to SIP RE-INVITE?
>>
>> Thanks.
>>
>>
>> --
>> _
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>
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Re: [asterisk-users] DTMF recognized after call establishment

[Asghar Mohammad]

i had this in past there was an ATA configured to send 9 at the end of
dialing in my case.


On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Hi,
>
> I am receiving DTMF without any reason after call establishment.
>
> The log as follows, and I suspect something related to directmedia,
> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
> making progress passing it to SIP/MAN-000a4b48
> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
> answered SIP/MAN-000a4b48
> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
> SIP/MyTrunk-000a4b49, duration 0 ms
> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
> '*' on SIP/MyTrunk-000a4b49
> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
> SIP/MyTrunk-000a4b49
> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
> SIP/MyTrunk-000a4b49, duration 0 ms
> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
> '8' on SIP/MyTrunk-000a4b49
> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
> SIP/MyTrunk-000a4b49
> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
> SIP/MAN-000a4af0, duration 100 ms
> [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
> duration 100 queued on SIP/MAN-000a4af0
> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued
> on SIP/MAN-000a4af0
> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
> SIP/MAN-000a4b41, duration 100 ms
> [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
> duration 100 queued on SIP/MAN-000a4b41
> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued
> on SIP/MAN-000a4b41
> [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
> (sip-trunk-inbound, 2127773456, 1) exited non-zero on
> 'SIP/MyTrunk-000a4af3'
> [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
> NoOp("SIP/MAN-000a4b09", "16") in new stack
> [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
> (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'
>
> Is this some thing related to SIP RE-INVITE?
>
> Thanks.
>
>
> --
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Re: [asterisk-users] DTMF Blips at end of Record() - 1.8.18

[James Lamanna]

On Wed, Feb 20, 2013 at 10:49 AM, James Lamanna  wrote:

> Hi,
> I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end
> the recording on the recording itself.
> Is there an easy way to truncate the last 200ms of the recording or so to
> eliminate this?
> The DTMF is coming in through rfc2833 and not inband.
>
>
I have another PBX running the same exact version of asterisk that's newer
that doesn't exhibit the same problem.
I'm wondering if it is a timing thing? The PBX with the issue seems to
respond slower to DTMF (Background() takes longer to get interrupted, etc..)
It is an older box so it is using res_timing_dahdi (dahdi_dummy) and the
newer box is using res_timing_timerfd.

Any suggestions would be welcome.

Thanks.

-- James
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Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone

[mohammad aliasgari]

The call is from an FXS phone connected to a DIGIUM tdm400 card, used 
by asterisk.when an FXS calls a SIP phone, DTMF detections are not 
displayed/logged in asterisk CLI console,Although I have enabled DTMF in logger 
and console verbose is 5.But wen SIP dials the FXS, and then presses DTMF 
digits, they are displayed and logged.what's missing?
 Regards


--- On Wed, 11/28/12, Joshua Colp  wrote:

From: Joshua Colp 
Subject: Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Wednesday, November 28, 2012, 1:10 PM

mohammad aliasgari wrote:
> Dear all,

Hola,

> having verbose level 5, and enabling dtmf logging in
> /etc/asterisk/logger.conf
> console => notice,warning,error,debug,dtmf
> I receive dtmf detected, in a SIP-PSTN call, as follows



> 
> Why don't I receive DTMF that are dialed by a PSTN phone?

How is the call from your PSTN phone being delivered into Asterisk?

If coming in over SIP the configuration may be incorrect.

Cheers,

-- Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone

[Joshua Colp]

mohammad aliasgari wrote:

Dear all,


Hola,


having verbose level 5, and enabling dtmf logging in
/etc/asterisk/logger.conf
console => notice,warning,error,debug,dtmf
I receive dtmf detected, in a SIP-PSTN call, as follows






Why don't I receive DTMF that are dialed by a PSTN phone?


How is the call from your PSTN phone being delivered into Asterisk?

If coming in over SIP the configuration may be incorrect.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] DTMF Payload Settings

[Joshua Colp]

Necati Demir wrote:

Hello,


Hola,


The service provider wants me to setup dtmfmode to rfc2833 and dtmf
payload to 101.

I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload?


Asterisk already uses payload 101 for RFC2833 so you should be fine with 
dtmfmode=rfc2833


Cheers,

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Re: [asterisk-users] DTMF inband with telephone-event in SDP

[Joshua Colp]

Jakob Hirsch wrote:

Hello everyone!


Hola,


We use Asterisk for various services like voicemail. Our SIP clients
usually use rtp events (rfc2833) for DTMF, which works just fine and
independent from the codec (g711 vs. g726 etc.).

Now we noticed there are some SIP clients that announce telephone-event
in their SDP, but send their DTMF inband. The problem with that is, that
Asterisk obviously does not try to detect inband DTMF after seeing the
telephone-event payload type in the SDP.


Generally DTMF is something that has to be configured on both sides, you 
can't just configure it on one and have the negotiation force it to be that.



So we are in a kind of dilemma:
- dtmfmode=auto (and dtmfmode=rfc2833) will work for most, but not for
the described ones.
- dtmfmode=inband would also work for most, but of course not for the
ones using g726 et al.

Is there any Asterisk setting to force inband DTMF detection (with
non-compressing codecs only, of course)? I browsed the code without result.


Unfortunately there isn't a way to force this as you describe out of the 
box, you would have to make changes to chan_sip or explicitly have the 
clients configured properly.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

Sorry the attachment was too big. here is link:
http://www.2shared.com/file/Ola640Pn/doubledigit.html


On Fri, Oct 12, 2012 at 9:24 AM, SamyGo  wrote:
> Why am I feeling like I'm the only one here who is not able to see any
> pastebin link or attachments in this thread !
>
>

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Re: [asterisk-users] DTMF digits are coming through twice

[SamyGo]

Why am I feeling like I'm the only one here who is not able to see any
pastebin link or attachments in this thread !

On Fri, Oct 12, 2012 at 6:18 PM, Vik Killa  wrote:

> The trace is attached 3 emails back.
>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

The trace is attached 3 emails back.

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Re: [asterisk-users] DTMF digits are coming through twice

[SamyGo]

Well like I said before and you increased my doubt as well by saying that
this happens from callers using E-link internet. Can you share the trace !

On Fri, Oct 12, 2012 at 5:24 PM, Vik Killa  wrote:

> Any ideas?
>
> On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa  wrote:
> > Call was to 7167436110
> >
>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

Any ideas?

On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa  wrote:
> Call was to 7167436110
>

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Re: [asterisk-users] DTMF digits are coming through twice

[SamyGo]

Can you share your pcap trace !

On Thu, Oct 11, 2012 at 5:16 PM, Vik Killa  wrote:

> Only callers calling from Earthlink internet connection
>
> On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly  wrote:
> > Is this happening for all callers, or just iPhone callers?
> >
> > --Don
> >
>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

Only callers calling from Earthlink internet connection

On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly  wrote:
> Is this happening for all callers, or just iPhone callers?
>
> --Don
>

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Re: [asterisk-users] DTMF digits are coming through twice

[Don Kelly]

Is this happening for all callers, or just iPhone callers?

--Don

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vik Killa
Sent: Wednesday, October 10, 2012 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF digits are coming through twice

I'm not sure I follow, the packet capture on the asterisk server shows
double digits being entered. Does that mean it's the source?

On Wed, Oct 10, 2012 at 11:55 AM, SamyGo  wrote:
> Hi,
>
> Not exactly a solution, but I'm sure you must've taken pcap traces of 
> a few such sample calls. See in their RTPs that you are receiving 
> repeatedly same RTPs which will tell you that any DTMF packet is 
> coming in twice by the source or not !
> just one such simple pcap will help you identify at whose end the 
> issue lies. If the source is your vendor sending you RTPs twice for 
> DTMFs then send them the capture and ask them to fix it however they can.
>
> BR,
> Sammy
>
> On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa  wrote:
>>
>> I've been running an Asterisk server (1.6.2.17.2) for over a year 
>> without any major issues. All of a sudden people are unable to login 
>> to their voicemail because Asterisk is seeing DTMF twice for each 
>> digit the caller pushes. We've noticed the problem only consistently 
>> happens to callers from specific locations. All the locations having 
>> the issue use te same ITSP and internet provider. The ITSP swears 
>> it's not them because they tested outside their phone system, 
>> straight from a d-mark. We've tinkered and played with all options in 
>> Asterisk related to DTMF with no success. Aside from upgrading 
>> Asterisk, I'm at a loss as to how to fix this. This system has worked 
>> for over a year without this issue and all of a sudden it appears. My 
>> thought is that it's the internet provider (Earthlink) but they say 
>> it can not possibly be them. Here is my DTMF settings in sip.conf:
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>>
>> Any input appreciated! Thanks.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
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>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

After comparing packet captures of good and bad calls. It looks like
the double digit is coming from rfc2833 and dtmf inband.   It looks
like the inband tone is splitting the rfc2833 in two? Is there some
way to resolve this???

On Wed, Oct 10, 2012 at 12:28 PM, Vik Killa  wrote:
> I'm not sure I follow, the packet capture on the asterisk server shows
> double digits being entered. Does that mean it's the source?
>
> On Wed, Oct 10, 2012 at 11:55 AM, SamyGo  wrote:
>> Hi,
>>
>> Not exactly a solution, but I'm sure you must've taken pcap traces of a few
>> such sample calls. See in their RTPs that you are receiving repeatedly same
>> RTPs which will tell you that any DTMF packet is coming in twice by the
>> source or not !
>> just one such simple pcap will help you identify at whose end the issue
>> lies. If the source is your vendor sending you RTPs twice for DTMFs then
>> send them the capture and ask them to fix it however they can.
>>
>> BR,
>> Sammy
>>
>> On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa  wrote:
>>>
>>> I've been running an Asterisk server (1.6.2.17.2) for over a year
>>> without any major issues. All of a sudden people are unable to login
>>> to their voicemail because Asterisk is seeing DTMF twice for each
>>> digit the caller pushes. We've noticed the problem only consistently
>>> happens to callers from specific locations. All the locations having
>>> the issue use te same ITSP and internet provider. The ITSP swears it's
>>> not them because they tested outside their phone system, straight from
>>> a d-mark. We've tinkered and played with all options in Asterisk
>>> related to DTMF with no success. Aside from upgrading Asterisk, I'm at
>>> a loss as to how to fix this. This system has worked for over a year
>>> without this issue and all of a sudden it appears. My thought is that
>>> it's the internet provider (Earthlink) but they say it can not
>>> possibly be them. Here is my DTMF settings in sip.conf:
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>>
>>> Any input appreciated! Thanks.
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>http://www.asterisk.org/hello
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

I'm not sure I follow, the packet capture on the asterisk server shows
double digits being entered. Does that mean it's the source?

On Wed, Oct 10, 2012 at 11:55 AM, SamyGo  wrote:
> Hi,
>
> Not exactly a solution, but I'm sure you must've taken pcap traces of a few
> such sample calls. See in their RTPs that you are receiving repeatedly same
> RTPs which will tell you that any DTMF packet is coming in twice by the
> source or not !
> just one such simple pcap will help you identify at whose end the issue
> lies. If the source is your vendor sending you RTPs twice for DTMFs then
> send them the capture and ask them to fix it however they can.
>
> BR,
> Sammy
>
> On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa  wrote:
>>
>> I've been running an Asterisk server (1.6.2.17.2) for over a year
>> without any major issues. All of a sudden people are unable to login
>> to their voicemail because Asterisk is seeing DTMF twice for each
>> digit the caller pushes. We've noticed the problem only consistently
>> happens to callers from specific locations. All the locations having
>> the issue use te same ITSP and internet provider. The ITSP swears it's
>> not them because they tested outside their phone system, straight from
>> a d-mark. We've tinkered and played with all options in Asterisk
>> related to DTMF with no success. Aside from upgrading Asterisk, I'm at
>> a loss as to how to fix this. This system has worked for over a year
>> without this issue and all of a sudden it appears. My thought is that
>> it's the internet provider (Earthlink) but they say it can not
>> possibly be them. Here is my DTMF settings in sip.conf:
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>>
>> Any input appreciated! Thanks.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] DTMF digits are coming through twice

[SamyGo]

Hi,

Not exactly a solution, but I'm sure you must've taken pcap traces of a few
such sample calls. See in their RTPs that you are receiving repeatedly same
RTPs which will tell you that any DTMF packet is coming in twice by the
source or not !
just one such simple pcap will help you identify at whose end the issue
lies. If the source is your vendor sending you RTPs twice for DTMFs then
send them the capture and ask them to fix it however they can.

BR,
Sammy

On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa  wrote:

> I've been running an Asterisk server (1.6.2.17.2) for over a year
> without any major issues. All of a sudden people are unable to login
> to their voicemail because Asterisk is seeing DTMF twice for each
> digit the caller pushes. We've noticed the problem only consistently
> happens to callers from specific locations. All the locations having
> the issue use te same ITSP and internet provider. The ITSP swears it's
> not them because they tested outside their phone system, straight from
> a d-mark. We've tinkered and played with all options in Asterisk
> related to DTMF with no success. Aside from upgrading Asterisk, I'm at
> a loss as to how to fix this. This system has worked for over a year
> without this issue and all of a sudden it appears. My thought is that
> it's the internet provider (Earthlink) but they say it can not
> possibly be them. Here is my DTMF settings in sip.conf:
> dtmfmode=rfc2833
> relaxdtmf=yes
>
> Any input appreciated! Thanks.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/15/2012 6:28 PM, Matthew Jordan wrote:
> - Original Message -
>> From: "Vladimir Mikhelson" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Saturday, September 15, 2012 1:11:14 PM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
>
>> Can you please quote the appropriate RFC here?  It looks like the UA
>> just uses the real time of an event for time stamping.  I still would
>> like to understand the logic and the math here as 121920+2400=124320,
>> not 122080 or 122240.
> Asterisk claims to follow RFC 2833 for DTMF events over RTP.  That is the RFC
> that I'll refer to here.
>
> Yes, your UA is incrementing the source timestamp for each packet it sends
> corresponding to the same event, and that is incorrect.  RFC 2833 defines the
> usage of the RTP timestamp and the duration field as follows in Section 3.4:
>
> Timestamp: The RTP timestamp reflects the measurement point for
>the current packet. The event duration described in Section
>3.5 extends forwards from that time. The receiver calculates
>jitter for RTCP receiver reports based on all packets with a
>given timestamp. Note: The jitter value should primarily be
>used as a means for comparing the reception quality between
>two users or two time-periods, not as an absolute measure.
>
> duration: Duration of this digit, in timestamp units. Thus, the
>event began at the instant identified by the RTP timestamp
>and has so far lasted as long as indicated by this parameter.
>The event may or may not have ended.
>
>For a sampling rate of 8000 Hz, this field is sufficient to
>express event durations of up to approximately 8 seconds.
>
> Section 3.6 specifically states how these two fields are used when an event
> spans more than a single packet:
>
> If an event continues for more than one period, the source generating
>the events should send a new event packet with the RTP timestamp
>value corresponding to the beginning of the event and the duration of
>the event increased correspondingly. (The RTP sequence number is
>incremented by one for each packet.) If there has been no new event
>in the last interval, the event SHOULD be retransmitted three times
>or until the next event is recognized. This ensures that the duration
>of the event can be recognized correctly even if the last packet for
>an event is lost.
>
> Why is the timestamp handled this way?  
>
> Consider the following RTP packets denoting a DTMF event:
>
> Seq No | Timestamp | Event | Duration | End of Event | 
>  7991  | 13280 |  1|   0  |0 |
>  7992  | 13280 |  1|  160 |0 |
>  7993  | 13280 |  1|  320 |0 |
>
> Why does the RFC require the timestamp to be the same for all three packets?
> Well, what would happen if the first two packets were dropped?  In that case,
> we'd still know how to construct the DTMF digit - we know the relative source
> time denoting the beginning of the DTMF event (13280), and we know the
> duration (320) - which is 40 ms.  So we can start the DTMF digit at our
> appropriate relative time, and we can continue it for the appropriate length
> of time.
>
> What would happen instead if the timestamp increased in each RTP packet?
>
> Seq No | Timestamp | Event | Duration | End of Event | 
>  7991  | 13280 |  1|   0  |0 |
>  7992  | 13440 |  1|  160 |0 |
>  7993  | 13600 |  1|  320 |0 |
>
> Now if the first two packets are dropped, we would begin generating the DTMF
> tone at the wrong time - we'd think it was supposed to start at the source
> relative time of t=13600, when it should have started at t=13280.  This could
> cause a delay in the tone generation by 40 ms.
>
> This answers two of your questions:
>
>>>   Because RTP packets can arrive out of order, Asterisk is using
>>> the timestamp to determine if the packets correspond to the same
>>> DTMF event.
>> Is that something mandated by the respective RFC?  In other words is
>> it
>> "MUST" or "MAY" per RFC?  Or is it not there at all?
> It happens because the RFC specifically says the timestamp should be the same
> for all event packets corresponding to the same event.  Its not really a MAY,
> MUST, SHOULD, or SHALL.  Its just how the RTP timestamp is supposed to be 
> used.
>
> I'll explain the out of order scenario I alluded to later.
>
>> Why does the retransmission happen at all?  Maybe something else is
>> broken?
> Its happening because th

Re: [asterisk-users] DTMF digits falsely detected

[Matthew Jordan]

- Original Message -
> From: "Vladimir Mikhelson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Saturday, September 15, 2012 1:11:14 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected



> Can you please quote the appropriate RFC here?  It looks like the UA
> just uses the real time of an event for time stamping.  I still would
> like to understand the logic and the math here as 121920+2400=124320,
> not 122080 or 122240.

Asterisk claims to follow RFC 2833 for DTMF events over RTP.  That is the RFC
that I'll refer to here.

Yes, your UA is incrementing the source timestamp for each packet it sends
corresponding to the same event, and that is incorrect.  RFC 2833 defines the
usage of the RTP timestamp and the duration field as follows in Section 3.4:

Timestamp: The RTP timestamp reflects the measurement point for
   the current packet. The event duration described in Section
   3.5 extends forwards from that time. The receiver calculates
   jitter for RTCP receiver reports based on all packets with a
   given timestamp. Note: The jitter value should primarily be
   used as a means for comparing the reception quality between
   two users or two time-periods, not as an absolute measure.

duration: Duration of this digit, in timestamp units. Thus, the
   event began at the instant identified by the RTP timestamp
   and has so far lasted as long as indicated by this parameter.
   The event may or may not have ended.

   For a sampling rate of 8000 Hz, this field is sufficient to
   express event durations of up to approximately 8 seconds.

Section 3.6 specifically states how these two fields are used when an event
spans more than a single packet:

If an event continues for more than one period, the source generating
   the events should send a new event packet with the RTP timestamp
   value corresponding to the beginning of the event and the duration of
   the event increased correspondingly. (The RTP sequence number is
   incremented by one for each packet.) If there has been no new event
   in the last interval, the event SHOULD be retransmitted three times
   or until the next event is recognized. This ensures that the duration
   of the event can be recognized correctly even if the last packet for
   an event is lost.

Why is the timestamp handled this way?  

Consider the following RTP packets denoting a DTMF event:

Seq No | Timestamp | Event | Duration | End of Event | 
 7991  | 13280 |  1|   0  |0 |
 7992  | 13280 |  1|  160 |0 |
 7993  | 13280 |  1|  320 |0 |

Why does the RFC require the timestamp to be the same for all three packets?
Well, what would happen if the first two packets were dropped?  In that case,
we'd still know how to construct the DTMF digit - we know the relative source
time denoting the beginning of the DTMF event (13280), and we know the
duration (320) - which is 40 ms.  So we can start the DTMF digit at our
appropriate relative time, and we can continue it for the appropriate length
of time.

What would happen instead if the timestamp increased in each RTP packet?

Seq No | Timestamp | Event | Duration | End of Event | 
 7991  | 13280 |  1|   0  |0 |
 7992  | 13440 |  1|  160 |0 |
 7993  | 13600 |  1|  320 |0 |

Now if the first two packets are dropped, we would begin generating the DTMF
tone at the wrong time - we'd think it was supposed to start at the source
relative time of t=13600, when it should have started at t=13280.  This could
cause a delay in the tone generation by 40 ms.

This answers two of your questions:

> 
> >   Because RTP packets can arrive out of order, Asterisk is using
> > the timestamp to determine if the packets correspond to the same
> > DTMF event.
> 
> Is that something mandated by the respective RFC?  In other words is
> it
> "MUST" or "MAY" per RFC?  Or is it not there at all?

It happens because the RFC specifically says the timestamp should be the same
for all event packets corresponding to the same event.  Its not really a MAY,
MUST, SHOULD, or SHALL.  Its just how the RTP timestamp is supposed to be used.

I'll explain the out of order scenario I alluded to later.

> Why does the retransmission happen at all?  Maybe something else is
> broken?

Its happening because the RFC in Section 3.6 says that when the DTMF end is
detected, that is, there are no new events detected in the last interval, the
event should be retransmitted three times.  In that regard, your UA is behaving
correctly (note in my previous e-mail that the duration did not increase) - but
the new RTP timestamps denote different times that the event began, which is
incorrect.

So, why did the handling of this change in Asterisk 1.8.16.0 and 10.8.0?

Asteri

Re: [asterisk-users] DTMF digits falsely detected

[Alec Davis]

> And just to make sure.  In both scenarios, normal digit press 
> and prolonged digit press, you did not reproduce the problem 
> we are discussing with X-Lite.  Is that correct?
> 

Correct, everything with X-Lite 3.0 and asterisk 1.8.16.0 worked correctly
with short, normal and long key presses while in voicemail.

Alec


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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/15/2012 5:16 PM, Alec Davis wrote:
>  
>>> [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end 
>> '4' received on
>>> SIP/alec-0009, duration 1660 
>>>
>>>
>>>
>> Alec,
>>
>> Interestingly in your log DTMF durations are even greater 
>> than in my original sampling.  Well, maybe my "duration" 
>> theory is not right.  But needed to exclude it first as it 
>> was on the surface.  I am assuming you dialed the digits 
>> normally, i.e. did not try to push a button longer than usual.
>>
> Yes I did press digits longer than usual, as you'd highlighted that duration
> may be an issue.
> Pressing digits normally and fast, gave the expected correct results.
>
> Alec
>
>
>
Alec,

Thank you for the clarification.

And just to make sure.  In both scenarios, normal digit press and
prolonged digit press, you did not reproduce the problem we are
discussing with X-Lite.  Is that correct?

Thank you,
Vladimir



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Re: [asterisk-users] DTMF digits falsely detected

[Alec Davis]

 
> > [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end 
> '4' received on
> > SIP/alec-0009, duration 1660 
> >
> >
> >
> Alec,
> 
> Interestingly in your log DTMF durations are even greater 
> than in my original sampling.  Well, maybe my "duration" 
> theory is not right.  But needed to exclude it first as it 
> was on the surface.  I am assuming you dialed the digits 
> normally, i.e. did not try to push a button longer than usual.
> 

Yes I did press digits longer than usual, as you'd highlighted that duration
may be an issue.
Pressing digits normally and fast, gave the expected correct results.

Alec


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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

Matt,

Please see my answers in-line.  Thank you for looking into the issue.

-Vladimir


On 9/15/2012 12:36 PM, Matthew Jordan wrote:
> - Original Message -
>> From: "Vladimir Mikhelson" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Saturday, September 15, 2012 11:41:23 AM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>> Please take a look at the case
>> https://issues.asterisk.org/jira/browse/ASTERISK-20424?actionOrder=asc
>> I uploaded the PCAP captured on the Soft Phone end and the RTP debug
>> log.
>>
>> I ran the "old" Soft Phone", dialed *98, then entered "430", the
>> application heard "444333000".
>>
> Thank you for uploading the log files.  It appears as if the UA sending
> the DTMF is incorrectly increasing the timestamp in the end event 
> retransmissions:
>
> [2012-09-15 11:02:49] VERBOSE[6392] res_rtp_asterisk.c: Got  RTP RFC2833 from 
>   172.17.135.5:13000 (type 101, seq 000231, ts 121920, len 04, mark 0, 
> event 0004, end 1, duration 02400) 
>
> [2012-09-15 11:02:49] VERBOSE[6392] res_rtp_asterisk.c: Got  RTP RFC2833 from 
>   172.17.135.5:13000 (type 101, seq 000232, ts 122080, len 04, mark 0, 
> event 0004, end 1, duration 02400) 
>
> [2012-09-15 11:02:49] VERBOSE[6392] res_rtp_asterisk.c: Got  RTP RFC2833 from 
>   172.17.135.5:13000 (type 101, seq 000233, ts 122240, len 04, mark 0, 
> event 0004, end 1, duration 02400) 
>
> You'll note that the timestamp is increasing in each subsequent 
> retransmission.
> The timestamp should be the same across all three packets with an increasing
> sequence number.

Can you please quote the appropriate RFC here?  It looks like the UA
just uses the real time of an event for time stamping.  I still would
like to understand the logic and the math here as 121920+2400=124320,
not 122080 or 122240.

Why does the retransmission happen at all?  Maybe something else is broken?


>   Because RTP packets can arrive out of order, Asterisk is using
> the timestamp to determine if the packets correspond to the same DTMF event.

Is that something mandated by the respective RFC?  In other words is it
"MUST" or "MAY" per RFC?  Or is it not there at all?

> The fact that both the sequence number and the timestamp are increasing would
> typically imply that the next end packet received is actually an out of order 
> packet
> for a subsequent DTMF digit.

See the question above.

>
> I'm not entirely sure what you mean by the following:
>
> "Another interesting thing which our friend Matt apparently did not pay
> attention to was the fact that dialing worked fine with Minipax, it was
> the applications where problems started.  Sounds like his latest patch
> was not applied consistently throughout the system.  Good news for now,
> but could change in the future."
>
> Dialing would not apply here, as a SIP device will not use RFC2833 DTMF to
> indicate the UA it wishes to establish a dialog with in an INVITE request (its
> in the INVITE request itself) - unless overlap dialing somehow became 
> involved.
> I doubt that's what you referring to here, however.

Sorry, my mistake.  Somehow I assumed that dialing was processed in
IVR.  Even in this case my statement would have been incorrect as IVR is
yet another application.

I need to test with IVR, but my bet results will be the same as with
Voice Mail.  At least that is how it sounds from your explanation of
what behavior was modified and where.

> This would only exhibit once the UA in question was actually sending DTMF
> over RTP.
>
> This patch only applied to decoding RFC2833 DTMF on the inbound read side of
> res_rtp_asterisk.  That's the only place it made sense to apply the handling
> for receiving out of order RFC2833 DTMF.  Outbound transmission would 
> obviously
> not be affected.  Similarly, devices communicating using chan_ooh323 - or even
> the majority of SIP devices - would not reproduce this, as they may be 
> transmitting
> the DTMF digits correctly.  I'm not sure how chan_dahdi would enter into this 
> at all,
> as its certainly not going to use the RTP engine.

As I did not know what exactly caused the DTMF recognition failure I
tested almost all channels I use, e.g. I called in via a POTS line.

> Again, thanks for uploading the files.  We'll take a look at this issue on
> Monday to see if there's anyway the behavior can be modified to account for
> the non-compliant endpoint while still handling out of order DTMF 
> transmission.

Once again please refer to the RFC you use as the specification.

> --
> Matthew Jordan
> Digiu

Re: [asterisk-users] DTMF digits falsely detected

[Matthew Jordan]

- Original Message -
> From: "Vladimir Mikhelson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Saturday, September 15, 2012 11:41:23 AM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
> Please take a look at the case
> https://issues.asterisk.org/jira/browse/ASTERISK-20424?actionOrder=asc
> I uploaded the PCAP captured on the Soft Phone end and the RTP debug
> log.
> 
> I ran the "old" Soft Phone", dialed *98, then entered "430", the
> application heard "444333000".
> 

Thank you for uploading the log files.  It appears as if the UA sending
the DTMF is incorrectly increasing the timestamp in the end event 
retransmissions:

[2012-09-15 11:02:49] VERBOSE[6392] res_rtp_asterisk.c: Got  RTP RFC2833 from   
172.17.135.5:13000 (type 101, seq 000231, ts 121920, len 04, mark 0, event 
0004, end 1, duration 02400) 

[2012-09-15 11:02:49] VERBOSE[6392] res_rtp_asterisk.c: Got  RTP RFC2833 from   
172.17.135.5:13000 (type 101, seq 000232, ts 122080, len 04, mark 0, event 
0004, end 1, duration 02400) 

[2012-09-15 11:02:49] VERBOSE[6392] res_rtp_asterisk.c: Got  RTP RFC2833 from   
172.17.135.5:13000 (type 101, seq 000233, ts 122240, len 04, mark 0, event 
0004, end 1, duration 02400) 

You'll note that the timestamp is increasing in each subsequent retransmission.
The timestamp should be the same across all three packets with an increasing
sequence number.  Because RTP packets can arrive out of order, Asterisk is using
the timestamp to determine if the packets correspond to the same DTMF event.
The fact that both the sequence number and the timestamp are increasing would
typically imply that the next end packet received is actually an out of order 
packet
for a subsequent DTMF digit.

I'm not entirely sure what you mean by the following:

"Another interesting thing which our friend Matt apparently did not pay
attention to was the fact that dialing worked fine with Minipax, it was
the applications where problems started.  Sounds like his latest patch
was not applied consistently throughout the system.  Good news for now,
but could change in the future."

Dialing would not apply here, as a SIP device will not use RFC2833 DTMF to
indicate the UA it wishes to establish a dialog with in an INVITE request (its
in the INVITE request itself) - unless overlap dialing somehow became involved.
I doubt that's what you referring to here, however.

This would only exhibit once the UA in question was actually sending DTMF
over RTP.

This patch only applied to decoding RFC2833 DTMF on the inbound read side of
res_rtp_asterisk.  That's the only place it made sense to apply the handling
for receiving out of order RFC2833 DTMF.  Outbound transmission would obviously
not be affected.  Similarly, devices communicating using chan_ooh323 - or even
the majority of SIP devices - would not reproduce this, as they may be 
transmitting
the DTMF digits correctly.  I'm not sure how chan_dahdi would enter into this 
at all,
as its certainly not going to use the RTP engine.

Again, thanks for uploading the files.  We'll take a look at this issue on
Monday to see if there's anyway the behavior can be modified to account for
the non-compliant endpoint while still handling out of order DTMF transmission.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

>> Hopefully the initial poster still has the configuration to 
>> produce the files for you.
>>
>> Are you saying the DTMF logs I attached do not provide enough 
>> evidence to support the theory of the DTMF length being the 
>> cause of this issue?
>>
>> -Vladimir
>>
> Vladimir,
>   What was the Softphone/Version you were using to get this to fail.
>
>   I'm using an old version of X-Lite, V3.0 build 56125 and with
> asterisk 1.8.16.0 when in voicemail I was unable to get any errors. DTMF log
> below.
>   
> [2012-09-15 22:36:39.974909] DTMF[1706] channel.c: DTMF begin '1' received
> on SIP/alec-0009
> [2012-09-15 22:36:39.974985] DTMF[1706] channel.c: DTMF begin ignored '1' on
> SIP/alec-0009
> [2012-09-15 22:36:40.514978] DTMF[1706] channel.c: DTMF end '1' received on
> SIP/alec-0009, duration 560  ms
> [2012-09-15 22:36:40.515037] DTMF[1706] channel.c: DTMF end passthrough '1'
> on SIP/alec-0009
> [2012-09-15 22:36:41.014955] DTMF[1706] channel.c: DTMF begin '2' received
> on SIP/alec-0009
> [2012-09-15 22:36:41.015009] DTMF[1706] channel.c: DTMF begin ignored '2' on
> SIP/alec-0009
> [2012-09-15 22:36:41.459045] DTMF[1706] channel.c: DTMF end '2' received on
> SIP/alec-0009, duration 460  ms
> [2012-09-15 22:36:41.459089] DTMF[1706] channel.c: DTMF end passthrough '2'
> on SIP/alec-0009
> [2012-09-15 22:36:41.909042] DTMF[1706] channel.c: DTMF begin '3' received
> on SIP/alec-0009
> [2012-09-15 22:36:41.909093] DTMF[1706] channel.c: DTMF begin ignored '3' on
> SIP/alec-0009
> [2012-09-15 22:36:42.429177] DTMF[1706] channel.c: DTMF end '3' received on
> SIP/alec-0009, duration 540  ms
> [2012-09-15 22:36:42.429236] DTMF[1706] channel.c: DTMF end passthrough '3'
> on SIP/alec-0009
> [2012-09-15 22:36:42.849091] DTMF[1706] channel.c: DTMF begin '4' received
> on SIP/alec-0009
> [2012-09-15 22:36:42.849185] DTMF[1706] channel.c: DTMF begin ignored '4' on
> SIP/alec-0009
> [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end '4' received on
> SIP/alec-0009, duration 1660 
>
>

Alec,

Please take a look at the case
https://issues.asterisk.org/jira/browse/ASTERISK-20424?actionOrder=asc  
I uploaded the PCAP captured on the Soft Phone end and the RTP debug log.

I ran the "old" Soft Phone", dialed *98, then entered "430", the
application heard "444333000".

Thank you,
Vladimir


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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/15/2012 6:16 AM, Alec Davis wrote:
>  
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
>> Vladimir Mikhelson
>> Sent: Saturday, 15 September 2012 5:56 p.m.
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>>
>> On 9/14/2012 11:04 PM, Matthew Jordan wrote:
>>> - Original Message -
>>>> From: "Vladimir Mikhelson" 
>>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>>>> 
>>>> Sent: Friday, September 14, 2012 10:39:30 PM
>>>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>>>
>>>>
>>>> On 9/14/2012 10:11 PM, Matthew Jordan wrote:
>>>>> ----- Original Message -
>>>>>> From: "Vladimir Mikhelson" 
>>>>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>>>>> 
>>>>>> Sent: Friday, September 14, 2012 9:24:41 PM
>>>>>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>>>>>
>>>>>>
>>>>>> On 9/14/2012 6:04 PM, Alec Davis wrote:
>>>>>>>> -Original Message-
>>>>>>>> From: asterisk-users-boun...@lists.digium.com
>>>>>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
>>>>>>>> Vieri
>>>>>>>> Sent: Saturday, 15 September 2012 8:45 a.m.
>>>>>>>> To: asterisk-users@lists.digium.com
>>>>>>>> Subject: [asterisk-users] DTMF digits falsely detected
>>>>>>>>
>>>>>> Can it be related to
>>>>>> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
>>>>>>
>>>>>> -Vladimir
>>>>> Most likely not.  If the SIP peer is using rfc2833 DTMF, its most 
>>>>> likely related to r370252.
>>>>>
>>>>> Please file an issue on the issue tracker, 
>>>>> https://issues.asterisk.org/jira.
>>>>> Please include a pcap of the RTP stream and a DEBUG log with RTP 
>>>>> debug enabled, using 'rtp set debug on'.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> --
>>>>> Matthew Jordan
>>>>>
>>>> Matt,
>>>>
>>>> I have created the issue.  See
>>>>
>> https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedComment
>>>> Id=197108#comment-197108
>>>>
>>>> Sorry, I will be unable to produce pcap and rtp debug as I 
>> have fixed 
>>>> the issue by uninstalling the Soft Phone I used for multiple years 
>>>> with no issues.
>>>>
>>>> -Vladimir
>>> Well, it'd be appreciated if someone who is experiencing 
>> this would be 
>>> willing to reproduce it and attach a pcap and DEBUG log to 
>> the issue.
>>> The bug fixed by that commit dealt with out of order DTMF; 
>> I suspect 
>>> that the problem is your soft phone is sending re-transmits 
>> of the end 
>>> event of the DTMF digit with an increasing timestamp.  The previous 
>>> behavior in Asterisk would most likely have been more 
>> tolerant of this 
>>> non-compliant scenario, but didn't handle the out of order 
>> packets as 
>>> well.
>>>
>>> Unfortunately, without evidence confirming that, there isn't much I 
>>> can do.
>>>
>>> --
>>> Matthew Jordan
>>>
>> Hopefully the initial poster still has the configuration to 
>> produce the files for you.
>>
>> Are you saying the DTMF logs I attached do not provide enough 
>> evidence to support the theory of the DTMF length being the 
>> cause of this issue?
>>
>> -Vladimir
>>
> Vladimir,
>   What was the Softphone/Version you were using to get this to fail.
>
>   I'm using an old version of X-Lite, V3.0 build 56125 and with
> asterisk 1.8.16.0 when in voicemail I was unable to get any errors. DTMF log
> below.
>   
> [2012-09-15 22:36:39.974909] DTMF[1706] channel.c: DTMF begin '1' received
> on SIP/alec-0009
> [2012-09-15 22:36:39.974985] DTMF[1706] channel.c: DTMF begin ignored '1' on
> SIP/alec-0009
> [2012-09-15 22:36:40.5

Re: [asterisk-users] DTMF digits falsely detected

[Alec Davis]

 

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> Vladimir Mikhelson
> Sent: Saturday, 15 September 2012 5:56 p.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
> 
> On 9/14/2012 11:04 PM, Matthew Jordan wrote:
> > - Original Message -
> >> From: "Vladimir Mikhelson" 
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> >> 
> >> Sent: Friday, September 14, 2012 10:39:30 PM
> >> Subject: Re: [asterisk-users] DTMF digits falsely detected
> >>
> >>
> >> On 9/14/2012 10:11 PM, Matthew Jordan wrote:
> >>> - Original Message -
> >>>> From: "Vladimir Mikhelson" 
> >>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >>>> 
> >>>> Sent: Friday, September 14, 2012 9:24:41 PM
> >>>> Subject: Re: [asterisk-users] DTMF digits falsely detected
> >>>>
> >>>>
> >>>> On 9/14/2012 6:04 PM, Alec Davis wrote:
> >>>>>> -Original Message-
> >>>>>> From: asterisk-users-boun...@lists.digium.com
> >>>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> >>>>>> Vieri
> >>>>>> Sent: Saturday, 15 September 2012 8:45 a.m.
> >>>>>> To: asterisk-users@lists.digium.com
> >>>>>> Subject: [asterisk-users] DTMF digits falsely detected
> >>>>>>
> >>>> Can it be related to
> >>>> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
> >>>>
> >>>> -Vladimir
> >>> Most likely not.  If the SIP peer is using rfc2833 DTMF, its most 
> >>> likely related to r370252.
> >>>
> >>> Please file an issue on the issue tracker, 
> >>> https://issues.asterisk.org/jira.
> >>> Please include a pcap of the RTP stream and a DEBUG log with RTP 
> >>> debug enabled, using 'rtp set debug on'.
> >>>
> >>> Thanks,
> >>>
> >>> --
> >>> Matthew Jordan
> >>>
> >> Matt,
> >>
> >> I have created the issue.  See
> >> 
> https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedComment
> >> Id=197108#comment-197108
> >>
> >> Sorry, I will be unable to produce pcap and rtp debug as I 
> have fixed 
> >> the issue by uninstalling the Soft Phone I used for multiple years 
> >> with no issues.
> >>
> >> -Vladimir
> > Well, it'd be appreciated if someone who is experiencing 
> this would be 
> > willing to reproduce it and attach a pcap and DEBUG log to 
> the issue.
> > The bug fixed by that commit dealt with out of order DTMF; 
> I suspect 
> > that the problem is your soft phone is sending re-transmits 
> of the end 
> > event of the DTMF digit with an increasing timestamp.  The previous 
> > behavior in Asterisk would most likely have been more 
> tolerant of this 
> > non-compliant scenario, but didn't handle the out of order 
> packets as 
> > well.
> >
> > Unfortunately, without evidence confirming that, there isn't much I 
> > can do.
> >
> > --
> > Matthew Jordan
> >
> Hopefully the initial poster still has the configuration to 
> produce the files for you.
> 
> Are you saying the DTMF logs I attached do not provide enough 
> evidence to support the theory of the DTMF length being the 
> cause of this issue?
> 
> -Vladimir
>

Vladimir,
What was the Softphone/Version you were using to get this to fail.

I'm using an old version of X-Lite, V3.0 build 56125 and with
asterisk 1.8.16.0 when in voicemail I was unable to get any errors. DTMF log
below.
  
[2012-09-15 22:36:39.974909] DTMF[1706] channel.c: DTMF begin '1' received
on SIP/alec-0009
[2012-09-15 22:36:39.974985] DTMF[1706] channel.c: DTMF begin ignored '1' on
SIP/alec-0009
[2012-09-15 22:36:40.514978] DTMF[1706] channel.c: DTMF end '1' received on
SIP/alec-0009, duration 560  ms
[2012-09-15 22:36:40.515037] DTMF[1706] channel.c: DTMF end passthrough '1'
on SIP/alec-0009
[2012-09-15 22:36:41.014955] DTMF[1706] channel.c: DTMF begin '2' received
on SIP/alec-0009
[2012-09-15 22:36:41.015009] DTMF[1706] channel.c: DTMF begin ignore

Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 11:04 PM, Matthew Jordan wrote:
> - Original Message -
>> From: "Vladimir Mikhelson" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Friday, September 14, 2012 10:39:30 PM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>>
>> On 9/14/2012 10:11 PM, Matthew Jordan wrote:
>>> - Original Message -
>>>> From: "Vladimir Mikhelson" 
>>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>>> 
>>>> Sent: Friday, September 14, 2012 9:24:41 PM
>>>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>>>
>>>>
>>>> On 9/14/2012 6:04 PM, Alec Davis wrote:
>>>>>> -Original Message-
>>>>>> From: asterisk-users-boun...@lists.digium.com
>>>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>>>>>> Vieri
>>>>>> Sent: Saturday, 15 September 2012 8:45 a.m.
>>>>>> To: asterisk-users@lists.digium.com
>>>>>> Subject: [asterisk-users] DTMF digits falsely detected
>>>>>>
>>>> Can it be related to
>>>> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
>>>>
>>>> -Vladimir
>>> Most likely not.  If the SIP peer is using rfc2833 DTMF, its most
>>> likely
>>> related to r370252.
>>>
>>> Please file an issue on the issue tracker,
>>> https://issues.asterisk.org/jira.
>>> Please include a pcap of the RTP stream and a DEBUG log with RTP
>>> debug
>>> enabled, using 'rtp set debug on'.
>>>
>>> Thanks,
>>>
>>> --
>>> Matthew Jordan
>>>
>> Matt,
>>
>> I have created the issue.  See
>> https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedCommentId=197108#comment-197108
>>
>> Sorry, I will be unable to produce pcap and rtp debug as I have fixed
>> the issue by uninstalling the Soft Phone I used for multiple years
>> with
>> no issues.
>>
>> -Vladimir
> Well, it'd be appreciated if someone who is experiencing this would be
> willing to reproduce it and attach a pcap and DEBUG log to the issue.
> The bug fixed by that commit dealt with out of order DTMF; I suspect
> that the problem is your soft phone is sending re-transmits of the end
> event of the DTMF digit with an increasing timestamp.  The previous
> behavior in Asterisk would most likely have been more tolerant of
> this non-compliant scenario, but didn't handle the out of order packets
> as well.
>
> Unfortunately, without evidence confirming that, there isn't much I can
> do.
>
> --
> Matthew Jordan
>
Hopefully the initial poster still has the configuration to produce the
files for you.

Are you saying the DTMF logs I attached do not provide enough evidence
to support the theory of the DTMF length being the cause of this issue?

-Vladimir


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Re: [asterisk-users] DTMF digits falsely detected

[Matthew Jordan]

- Original Message -
> From: "Vladimir Mikhelson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, September 14, 2012 10:39:30 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
> 
> On 9/14/2012 10:11 PM, Matthew Jordan wrote:
> >
> > - Original Message -
> >> From: "Vladimir Mikhelson" 
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >> 
> >> Sent: Friday, September 14, 2012 9:24:41 PM
> >> Subject: Re: [asterisk-users] DTMF digits falsely detected
> >>
> >>
> >> On 9/14/2012 6:04 PM, Alec Davis wrote:
> >>>> -Original Message-
> >>>> From: asterisk-users-boun...@lists.digium.com
> >>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> >>>> Vieri
> >>>> Sent: Saturday, 15 September 2012 8:45 a.m.
> >>>> To: asterisk-users@lists.digium.com
> >>>> Subject: [asterisk-users] DTMF digits falsely detected
> >>>>
> >> Can it be related to
> >> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
> >>
> >> -Vladimir
> > Most likely not.  If the SIP peer is using rfc2833 DTMF, its most
> > likely
> > related to r370252.
> >
> > Please file an issue on the issue tracker,
> > https://issues.asterisk.org/jira.
> > Please include a pcap of the RTP stream and a DEBUG log with RTP
> > debug
> > enabled, using 'rtp set debug on'.
> >
> > Thanks,
> >
> > --
> > Matthew Jordan
> >
> 
> Matt,
> 
> I have created the issue.  See
> https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedCommentId=197108#comment-197108
> 
> Sorry, I will be unable to produce pcap and rtp debug as I have fixed
> the issue by uninstalling the Soft Phone I used for multiple years
> with
> no issues.
> 
> -Vladimir

Well, it'd be appreciated if someone who is experiencing this would be
willing to reproduce it and attach a pcap and DEBUG log to the issue.
The bug fixed by that commit dealt with out of order DTMF; I suspect
that the problem is your soft phone is sending re-transmits of the end
event of the DTMF digit with an increasing timestamp.  The previous
behavior in Asterisk would most likely have been more tolerant of
this non-compliant scenario, but didn't handle the out of order packets
as well.

Unfortunately, without evidence confirming that, there isn't much I can
do.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 10:11 PM, Matthew Jordan wrote:
>
> - Original Message -
>> From: "Vladimir Mikhelson" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Friday, September 14, 2012 9:24:41 PM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>>
>> On 9/14/2012 6:04 PM, Alec Davis wrote:
>>>> -Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com
>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>>>> Vieri
>>>> Sent: Saturday, 15 September 2012 8:45 a.m.
>>>> To: asterisk-users@lists.digium.com
>>>> Subject: [asterisk-users] DTMF digits falsely detected
>>>>
>> Can it be related to
>> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
>>
>> -Vladimir
> Most likely not.  If the SIP peer is using rfc2833 DTMF, its most likely
> related to r370252.
>
> Please file an issue on the issue tracker, https://issues.asterisk.org/jira.
> Please include a pcap of the RTP stream and a DEBUG log with RTP debug
> enabled, using 'rtp set debug on'.
>
> Thanks,
>
> --
> Matthew Jordan
>

Matt,

I have created the issue.  See
https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedCommentId=197108#comment-197108

Sorry, I will be unable to produce pcap and rtp debug as I have fixed
the issue by uninstalling the Soft Phone I used for multiple years with
no issues.

-Vladimir


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Re: [asterisk-users] DTMF digits falsely detected

[Matthew Jordan]

- Original Message -
> From: "Vladimir Mikhelson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, September 14, 2012 9:24:41 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
> 
> On 9/14/2012 6:04 PM, Alec Davis wrote:
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> >> Vieri
> >> Sent: Saturday, 15 September 2012 8:45 a.m.
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] DTMF digits falsely detected
> >>
> 
> Can it be related to
> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
> 
> -Vladimir

Most likely not.  If the SIP peer is using rfc2833 DTMF, its most likely
related to r370252.

Please file an issue on the issue tracker, https://issues.asterisk.org/jira.
Please include a pcap of the RTP stream and a DEBUG log with RTP debug
enabled, using 'rtp set debug on'.

Thanks,

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 6:04 PM, Alec Davis wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
>> Sent: Saturday, 15 September 2012 8:45 a.m.
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
>> Hi,
>>
>> I have a context that basically does:
>>
>> Wait(1)
>> Background(message)
>> WaitExten(10)
>>
>> _6XX,1,DoSomething
>>
>> The problem is that when I reach this context and press some 
>> digits (eg. 6566604) then I can see in the log that Asterisk 
>> reads 6655666.
>> So it's actually reading the digits twice.
>> How can I avoid this?
>> Incoming channel type is ISDN (mISDN).
>>
> Are you saying every digit twice, or some digits twice.
> Where is the call originating from, GSM cell phone or landline?
>
> Which version of asterisk are you using?
>
> Alec Davis
>
>
Alec,

Can it be related to
https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??

-Vladimir
 

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 6:04 PM, Alec Davis wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
>> Sent: Saturday, 15 September 2012 8:45 a.m.
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
>> Hi,
>>
>> I have a context that basically does:
>>
>> Wait(1)
>> Background(message)
>> WaitExten(10)
>>
>> _6XX,1,DoSomething
>>
>> The problem is that when I reach this context and press some 
>> digits (eg. 6566604) then I can see in the log that Asterisk 
>> reads 6655666.
>> So it's actually reading the digits twice.
>> How can I avoid this?
>> Incoming channel type is ISDN (mISDN).
>>
> Are you saying every digit twice, or some digits twice.
> Where is the call originating from, GSM cell phone or landline?
>
> Which version of asterisk are you using?
>
> Alec Davis
>
>
>
Hi,

Started seeing similar abnormality today after the 1.8.16.0 upgrade.

In my case digits were repeated 3 times.

Several observations:

  * The problem only manifested itself on SIP channel.  OOH323 and DAHDI
did not exhibit this problem.
  * I was able to dial the extension with no issues, the problem started
in the Voice Mail application.

For example, in the case below I entered "430#"

[2012-09-14 11:50:06] VERBOSE[32019] app_read.c: -- User entered
'444333000'

Following is the excerpt from the DTMF log:

[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin '4' received on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin ignored '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end '4' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end passthrough '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end '4' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end passthrough '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end '4' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end passthrough '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin '3' received on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin ignored '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '3' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '3' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '3' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin '0' received on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin ignored '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '0' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '0' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '0' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin '#' received on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin ignored '#' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '#' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin emulation of '#'
with duration 300 queued on SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:07] DTMF[32019] channel.c: DTMF end emulation of '#'
queued on SIP/462-
[2012-09-14 11:50:07] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:07] DTMF[32019] channel.c: DTMF begin emulation of '#'
with duration 300 queued on SIP/462-

As a comparison here is an excerpt from the DTMF log of a similar call
from the same extension before the upgrade (Asterisk 1.8.15.1):

[2

Re: [asterisk-users] DTMF digits falsely detected

[Alec Davis]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
> Sent: Saturday, 15 September 2012 8:45 a.m.
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] DTMF digits falsely detected
> 
> Hi,
> 
> I have a context that basically does:
> 
> Wait(1)
> Background(message)
> WaitExten(10)
> 
> _6XX,1,DoSomething
> 
> The problem is that when I reach this context and press some 
> digits (eg. 6566604) then I can see in the log that Asterisk 
> reads 6655666.
> So it's actually reading the digits twice.
> How can I avoid this?
> Incoming channel type is ISDN (mISDN).
> 

Are you saying every digit twice, or some digits twice.
Where is the call originating from, GSM cell phone or landline?

Which version of asterisk are you using?

Alec Davis


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Re: [asterisk-users] DTMF Issue.

[Luis H. Forchesatto]

Up?

2012/8/20 Luis H. Forchesatto 

> Thanks for your answer.
>
> The logs where posted at pastebin, here the links:
>
> - Working Phone: http://pastebin.com/q3pHcwna
> - Not working phone: http://pastebin.com/iiCHPMmn
>
>
> 2012/8/20 Rusty Newton 
>
>> On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
>>
>>> Hi
>>>
>>> I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of
>>> ATA on the network who autenticate the phones: Linksys PAP2, Overtek
>>> OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at
>>> the same network all with g729 codecs and rfc2833 for the DTMF. Making
>>> calls via the Overtek ATA the DTMF works fine but at the others ATA it
>>> doesn't.
>>>
>>> My config:
>>>
>>> - asterisk 1.6.2.13
>>> - dahdi 2.3.0.1
>>> - The phones connected are all physical phones
>>>
>> There is additional data you can provide to make it easier for others to
>> help out:
>> If you can pastebin an Asterisk log including all message types plus
>> VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would
>> be very helpful.
>> A step beyond that is to also provide a SIP and RTP packet trace so that
>> whoever wants to help can look through it in Wireshark.
>>
>> If you can get the packet trace for the same calls you gather log data
>> for, that would be best.
>>
>> Thanks!
>>
>> [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+**
>> Debug+Information
>>
>> --
>> Rusty Newton
>> Digium, Inc | Open Source Community Support Manager
>> Check us out at: www.digium.com www.asterisk.org
>>
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>> To UNSUBSCRIBE or update options visit:
>>   
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
>
> --
> Att.*
> ***
>



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Re: [asterisk-users] DTMF Issue.

[Luis H. Forchesatto]

Thanks for your answer.

The logs where posted at pastebin, here the links:

- Working Phone: http://pastebin.com/q3pHcwna
- Not working phone: http://pastebin.com/iiCHPMmn

2012/8/20 Rusty Newton 

> On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
>
>> Hi
>>
>> I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of
>> ATA on the network who autenticate the phones: Linksys PAP2, Overtek
>> OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at
>> the same network all with g729 codecs and rfc2833 for the DTMF. Making
>> calls via the Overtek ATA the DTMF works fine but at the others ATA it
>> doesn't.
>>
>> My config:
>>
>> - asterisk 1.6.2.13
>> - dahdi 2.3.0.1
>> - The phones connected are all physical phones
>>
> There is additional data you can provide to make it easier for others to
> help out:
> If you can pastebin an Asterisk log including all message types plus
> VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would
> be very helpful.
> A step beyond that is to also provide a SIP and RTP packet trace so that
> whoever wants to help can look through it in Wireshark.
>
> If you can get the packet trace for the same calls you gather log data
> for, that would be best.
>
> Thanks!
>
> [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+**
> Debug+Information
>
> --
> Rusty Newton
> Digium, Inc | Open Source Community Support Manager
> Check us out at: www.digium.com www.asterisk.org
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>



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Re: [asterisk-users] DTMF Issue.

[Rusty Newton]

On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:

Hi

I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of 
ATA on the network who autenticate the phones: Linksys PAP2, 
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the 
VoIP server at the same network all with g729 codecs and rfc2833 for 
the DTMF. Making calls via the Overtek ATA the DTMF works fine but at 
the others ATA it doesn't.


My config:

- asterisk 1.6.2.13
- dahdi 2.3.0.1
- The phones connected are all physical phones
There is additional data you can provide to make it easier for others to 
help out:
If you can pastebin an Asterisk log including all message types plus 
VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that 
would be very helpful.
A step beyond that is to also provide a SIP and RTP packet trace so that 
whoever wants to help can look through it in Wireshark.


If you can get the packet trace for the same calls you gather log data 
for, that would be best.


Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org


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Re: [asterisk-users] DTMF transmission problem

[Noah Engelberth]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Noah Engelberth
> Sent: Thursday, August 02, 2012 1:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF transmission problem
> 
> 
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Noah Engelberth
> > Sent: Thursday, August 02, 2012 12:27 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] DTMF transmission problem
> >
> > > -Original Message-
> > > From: asterisk-users-boun...@lists.digium.com
> > > [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun
> > > Ruffell
> > > Sent: Thursday, August 02, 2012 11:06 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] DTMF transmission problem
> > >
> > > On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote:
> > > > I am having difficulties with customer-bound DTMF being very short
> > > > & clipped off (and basically unusable, as systems on the customer
> > > > side aren't recognizing the DTMF digits, and I can barely tell
> > > > that DTMF is there when I listen on a handset).
> > > >
> > > > My system set up as follows:
> > > >
> > > > PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE
> > >
> > > [snip]
> > >
> > > > ... Vocal call  quality is fine, DTMF is fine from the customer to
> > > > the PSTN, but DTMF from the PSTN to the customer isn't ...
> > >
> > >  [snip]
> > >
> > > > The same symptoms persist whether the PSTN or the CPE initiate the
> call.
> > >
> > > What is the dtmf mode of Metaswitch in the above diagram? Is it
> > > possible that it's muting the DTMF and then not generating the
> > > corresponding DTMF event messages?  Everytime I've seen "clipped"
> > > DTMF in the past it was due to imperfect muting at the PSTN -> SIP
> > interface.
> >
> > According to the gentleman that manages the Metaswitch, it's set to
> > allow for either in or out of band dtmf.  Based on the packet trace,
> > the packets are coming across as RFC 2833 RTP events.  Aside from the
> > very first digit, which Wireshark shows as 7 "RTP Event" packets and 3
> > "RTP Event (end)" packets, all the other ones on my test call came
> > across as 8 "RTP Event" packets and 3 "RTP Event (end)" packets.  All
> > of the RTP Event packets are in sequence for the call's RTP stream.
> >
> > Also, when I'm monitoring in Asterisk, if I configure logger.conf to
> > output DTMF events into the console, Asterisk is recognizing the DTMF:
> >
> > [Aug  2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4'
> > received on SIP/PSTN-SIP-PEER [Aug  2 12:25:25] DTMF[19319]:
> > channel.c:4146 __ast_read: DTMF begin passthrough '4' on SIP/
> > PSTN-SIP- PEER [Aug  2 12:25:25] DTMF[19319]: channel.c:4051
> > __ast_read: DTMF end '4' received on SIP/ PSTN-SIP-PEER, duration 280
> > ms [Aug  2 12:25:25]
> > DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted with begin
> '4'
> > on SIP/ PSTN-SIP-PEER [Aug  2 12:25:25] DTMF[19319]: channel.c:4120
> > __ast_read: DTMF end passthrough '4' on SIP/ PSTN-SIP-PEER
> >
> 
> Additional information I discovered after my previous reply:
> 
> I have a separate Asterisk VM instance (in all other ways the same
> implementation as above) that is running an IVR.  This instance has no issues
> with inbound DTMF within the IVR, but does exhibit the same symptoms for
> DTMF when bridged through to an IAX2 peer with the same settings as the
> first Asterisk VM.  On the second Asterisk (with the IVR), DTMF to my
> Cisco/Linksys SPA942 SIP phones works properly, but not to the IAX or SIP
> ATAs that I am using (the same ones I'm having problems with on the first
> Asterisk).  All of the live customers on the first Asterisk are ATAs, so I 
> don't
> know as of this time whether or not SPA phones are working correctly on the
> first server, though it's reasonable to assume they are.
> 
> In addition, calls from an SPA942 phone to the IAX or SIP ATAs are also not
> transmitting DTMF 

Re: [asterisk-users] DTMF transmission problem

[Noah Engelberth]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Noah Engelberth
> Sent: Thursday, August 02, 2012 12:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF transmission problem
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Shaun Ruffell
> > Sent: Thursday, August 02, 2012 11:06 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] DTMF transmission problem
> >
> > On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote:
> > > I am having difficulties with customer-bound DTMF being very short &
> > > clipped off (and basically unusable, as systems on the customer side
> > > aren't recognizing the DTMF digits, and I can barely tell that DTMF
> > > is there when I listen on a handset).
> > >
> > > My system set up as follows:
> > >
> > > PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE
> >
> > [snip]
> >
> > > ... Vocal call  quality is fine, DTMF is fine from the customer to
> > > the PSTN, but DTMF from the PSTN to the customer isn't ...
> >
> >  [snip]
> >
> > > The same symptoms persist whether the PSTN or the CPE initiate the call.
> >
> > What is the dtmf mode of Metaswitch in the above diagram? Is it
> > possible that it's muting the DTMF and then not generating the
> > corresponding DTMF event messages?  Everytime I've seen "clipped"
> > DTMF in the past it was due to imperfect muting at the PSTN -> SIP
> interface.
> 
> According to the gentleman that manages the Metaswitch, it's set to allow
> for either in or out of band dtmf.  Based on the packet trace, the packets are
> coming across as RFC 2833 RTP events.  Aside from the very first digit, which
> Wireshark shows as 7 "RTP Event" packets and 3 "RTP Event (end)" packets,
> all the other ones on my test call came across as 8 "RTP Event" packets and 3
> "RTP Event (end)" packets.  All of the RTP Event packets are in sequence for
> the call's RTP stream.
> 
> Also, when I'm monitoring in Asterisk, if I configure logger.conf to output
> DTMF events into the console, Asterisk is recognizing the DTMF:
> 
> [Aug  2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4'
> received on SIP/PSTN-SIP-PEER [Aug  2 12:25:25] DTMF[19319]:
> channel.c:4146 __ast_read: DTMF begin passthrough '4' on SIP/ PSTN-SIP-
> PEER [Aug  2 12:25:25] DTMF[19319]: channel.c:4051 __ast_read: DTMF end
> '4' received on SIP/ PSTN-SIP-PEER, duration 280 ms [Aug  2 12:25:25]
> DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted with begin '4'
> on SIP/ PSTN-SIP-PEER [Aug  2 12:25:25] DTMF[19319]: channel.c:4120
> __ast_read: DTMF end passthrough '4' on SIP/ PSTN-SIP-PEER
> 

Additional information I discovered after my previous reply:

I have a separate Asterisk VM instance (in all other ways the same 
implementation as above) that is running an IVR.  This instance has no issues 
with inbound DTMF within the IVR, but does exhibit the same symptoms for DTMF 
when bridged through to an IAX2 peer with the same settings as the first 
Asterisk VM.  On the second Asterisk (with the IVR), DTMF to my Cisco/Linksys 
SPA942 SIP phones works properly, but not to the IAX or SIP ATAs that I am 
using (the same ones I'm having problems with on the first Asterisk).  All of 
the live customers on the first Asterisk are ATAs, so I don't know as of this 
time whether or not SPA phones are working correctly on the first server, 
though it's reasonable to assume they are.

In addition, calls from an SPA942 phone to the IAX or SIP ATAs are also not 
transmitting DTMF to the ATA device's endpoint.  DTMF from the ATA device's 
endpoint to the SPA942 is working correctly, as is both directions of voice 
audio.

> >
> > You should be able to take a packet trace on the interface of the
> > Asterisk server communicating with the Metaswitch to determine whether
> > the problem first appears at the switch or in your Asterisk server.
> >
> > Cheers,
> > Shaun
> >
> > --
> > Shaun Ruffell
> > Digium, Inc. | Linux Kernel Developer
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
> > www.digium.com & www.asterisk.org
> >
> > --

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Re: [asterisk-users] DTMF transmission problem

[Noah Engelberth]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Shaun Ruffell
> Sent: Thursday, August 02, 2012 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF transmission problem
> 
> On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote:
> > I am having difficulties with customer-bound DTMF being very short &
> > clipped off (and basically unusable, as systems on the customer side
> > aren't recognizing the DTMF digits, and I can barely tell that DTMF is
> > there when I listen on a handset).
> >
> > My system set up as follows:
> >
> > PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE
> 
> [snip]
> 
> > ... Vocal call  quality is fine, DTMF is fine from the customer to the
> > PSTN, but DTMF from the PSTN to the customer isn't ...
> 
>  [snip]
> 
> > The same symptoms persist whether the PSTN or the CPE initiate the call.
> 
> What is the dtmf mode of Metaswitch in the above diagram? Is it possible
> that it's muting the DTMF and then not generating the corresponding DTMF
> event messages?  Everytime I've seen "clipped"
> DTMF in the past it was due to imperfect muting at the PSTN -> SIP interface.

According to the gentleman that manages the Metaswitch, it's set to allow for 
either in or out of band dtmf.  Based on the packet trace, the packets are 
coming across as RFC 2833 RTP events.  Aside from the very first digit, which 
Wireshark shows as 7 "RTP Event" packets and 3 "RTP Event (end)" packets, all 
the other ones on my test call came across as 8 "RTP Event" packets and 3 "RTP 
Event (end)" packets.  All of the RTP Event packets are in sequence for the 
call's RTP stream.

Also, when I'm monitoring in Asterisk, if I configure logger.conf to output 
DTMF events into the console, Asterisk is recognizing the DTMF:

[Aug  2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4' 
received on SIP/PSTN-SIP-PEER
[Aug  2 12:25:25] DTMF[19319]: channel.c:4146 __ast_read: DTMF begin 
passthrough '4' on SIP/ PSTN-SIP-PEER
[Aug  2 12:25:25] DTMF[19319]: channel.c:4051 __ast_read: DTMF end '4' received 
on SIP/ PSTN-SIP-PEER, duration 280 ms
[Aug  2 12:25:25] DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted 
with begin '4' on SIP/ PSTN-SIP-PEER
[Aug  2 12:25:25] DTMF[19319]: channel.c:4120 __ast_read: DTMF end passthrough 
'4' on SIP/ PSTN-SIP-PEER

> 
> You should be able to take a packet trace on the interface of the Asterisk
> server communicating with the Metaswitch to determine whether the
> problem first appears at the switch or in your Asterisk server.
> 
> Cheers,
> Shaun
> 
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
> www.digium.com & www.asterisk.org
> 
> --
> __
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] DTMF transmission problem

[Shaun Ruffell]

On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote:
> I am having difficulties with customer-bound DTMF being very short
> & clipped off (and basically unusable, as systems on the customer
> side aren't recognizing the DTMF digits, and I can barely tell
> that DTMF is there when I listen on a handset).
> 
> My system set up as follows:
> 
> PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE

[snip]

> ... Vocal call  quality is fine, DTMF is fine from the customer to the PSTN, 
> but
> DTMF from the PSTN to the customer isn't ...

 [snip]

> The same symptoms persist whether the PSTN or the CPE initiate the call.

What is the dtmf mode of Metaswitch in the above diagram? Is it
possible that it's muting the DTMF and then not generating the
corresponding DTMF event messages?  Everytime I've seen "clipped"
DTMF in the past it was due to imperfect muting at the PSTN -> SIP
interface.

You should be able to take a packet trace on the interface of the
Asterisk server communicating with the Metaswitch to determine
whether the problem first appears at the switch or in your Asterisk
server.

Cheers,
Shaun

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] DTMF forwarding and Page [SOLVED] [PATCH 1/1]

[Matteo Fortini]

Nevermind,
I checked the code, and A* is not using the "F" option in MeetMe for 
Page(), so it's not working by default.

Attached is a patch which fixes the problem for me, if anyone needs it.

Matteo

Il 11/02/2012 13:53, Matteo Fortini ha scritto:

Noone knows that? Where/whom could I ask?

Thanks

Il 10/02/2012 12:30, Matteo Fortini ha scritto:

Hi,
I'd like to implement some way of controlling remote SIP clients 
while in a call, to execute remote commands.


The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is 
connected to the previous one


I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().


TIA,
Matteo
diff -Nurd asterisk-1.6.2.20.orig/apps/app_page.c asterisk-1.6.2.20/apps/app_page.c
--- asterisk-1.6.2.20.orig/apps/app_page.c	2009-01-25 14:35:48.0 +0100
+++ asterisk-1.6.2.20/apps/app_page.c	2012-02-13 13:47:53.509396266 +0100
@@ -164,7 +164,7 @@
 		timeout = atoi(args.timeout);
 	}
 
-	snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
+	snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxFdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
 		(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 
 	/* Count number of extensions in list by number of ampersands + 1 */
@@ -247,7 +247,7 @@
 	}
 
 	if (!res) {
-		snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
+		snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqFxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
 			(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 		pbx_exec(chan, app, meetmeopts);
 	}
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Re: [asterisk-users] DTMF forwarding and Page

[Matteo Fortini]

Noone knows that? Where/whom could I ask?

Thanks

Il 10/02/2012 12:30, Matteo Fortini ha scritto:

Hi,
I'd like to implement some way of controlling remote SIP clients while 
in a call, to execute remote commands.


The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is 
connected to the previous one


I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().


TIA,
Matteo


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Re: [asterisk-users] DTMF Testing software to test IVR system

[Sammy Govind]

o in that case you need to observer the call flow in Server-B, i.e what is
the length of sound file playing. what DTMF it requires etc etc and once
you detect the call flow for a successful IVR traversal then mimic the
behaviour of the call from Server-A.
Thats all you can do.
Think of it exactly the same as Answering Machine Detection Algorithm, but
in your case its like Server-B Detection Algorithm :)

--
Regards,
Sammy

On Thu, Dec 29, 2011 at 2:15 PM, virendra bhati  wrote:

> In server B if I use SendDTMF then it means I am changing programming at
> server B. Actually I don't have right or permission to change programming
> in server B.
>
> otherwise your suggestion is best for channel base communication.
>
>
>
>
> On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind  wrote:
>
>> Easy, use Read() to capture the incoming DTMF from Server-B
>>
>> Server-A <> Server-B
>> Initiate-Call -> AnswerCall()
>> SendDTMF(5)--> Read()
>> Read()<-SendDTMF(4)
>> SendDTMF(3)--> Read()
>> Read()<-SendDTMF(2)
>> SendDTMF(1)--> Read()
>>
>>
>> Put proper GOTOIFs after reads if you like.
>>
>> --
>> Regards,
>> Sammy
>>
>> On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati wrote:
>>
>>> I originate calls from .call file and 1 channel I have at A server A and
>>> another channel at B server.
>>>
>>> *A server code is below:-*
>>>
>>> exten => 43689956,1,Answer()
>>> same => n,Wait(5)
>>> same => n,SendDTMF(1)
>>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>>> same => n,wait(2)
>>> same => n,SendDTMF(123456789012345#)
>>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>>> same => n,Hangup()
>>>
>>>  _  _
>>> |  A server  |  ___DTMF Send_=> | B server   |
>>> |_|  <=--- Responce -   |_|
>>>
>>> *B server code is below:-*
>>> At B server call come to 201 extension which is mention here..
>>>
>>> exten => _20[1-6],1,Answer()
>>> same => n,Ringing()
>>> same => n,wait(2)
>>> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
>>> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
>>> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
>>> $[${EXTEN}=205] ||
>>> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
>>> same => n,Hangup()
>>>
>>> Now I can send the DTMF from A to B. But How I will get the responce at
>>> server A. I checked all the channels variable but they didn't reply status
>>> of B server channel. All information I will get of server A. Main problem
>>> is that control reach to AGI and then I don't have any rights to do any
>>> update or modification on AGI. So if I can work on request and responce
>>> then it will be the last solution as per my knowledge.
>>>
>>> Is this possible with the dialplan or I am just westing time?
>>>
>>>
>>>
>>> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger 
>>> wrote:
>>>
 On 11-12-28 03:25 AM, virendra bhati wrote:

> Hi list,
>
> Is there any way in asterisk by which I make a call from server and
> then
> dialplan(IVR system) gets DTMF from it. I mean to say that
> automatically
> DTMF is sended by channels as per user defined,
>
> I read there is an application sendDTMF but I don't know how we can
> used it?
>
> like A script make the call by using localdail, .call file or any
> method.
> And after landing the call we send dtmf to IVR system automatically as
> per
> my script..
>
>
> *extensions.conf:-*
>
>
> exten =>  1234,1,Answer()
>  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
> rHelp,1,,10)
>  same =>  n,NoOp(${value})
>  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
>  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
>  same =>  n,Hangup()
>
> exten=>  support,1,Answer()
>  same =>  n,NoOp(you are at support section)
>  same =>  n,Hangup()
>
> exten=>  help,1,Answer()
>  same =>  n,NoOp(you are at help section)
>  same =>  n,Hangup()
>
>  We have DTMF based tests for the testsuite[1] that you could use.

 [1] 
 http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com & http://asterisk.org


 --
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 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live 

Re: [asterisk-users] DTMF Testing software to test IVR system

[virendra bhati]

In server B if I use SendDTMF then it means I am changing programming at
server B. Actually I don't have right or permission to change programming
in server B.

otherwise your suggestion is best for channel base communication.



On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind  wrote:

> Easy, use Read() to capture the incoming DTMF from Server-B
>
> Server-A <> Server-B
> Initiate-Call -> AnswerCall()
> SendDTMF(5)--> Read()
> Read()<-SendDTMF(4)
> SendDTMF(3)--> Read()
> Read()<-SendDTMF(2)
> SendDTMF(1)--> Read()
>
>
> Put proper GOTOIFs after reads if you like.
>
> --
> Regards,
> Sammy
>
> On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati wrote:
>
>> I originate calls from .call file and 1 channel I have at A server A and
>> another channel at B server.
>>
>> *A server code is below:-*
>>
>> exten => 43689956,1,Answer()
>> same => n,Wait(5)
>> same => n,SendDTMF(1)
>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>> same => n,wait(2)
>> same => n,SendDTMF(123456789012345#)
>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>> same => n,Hangup()
>>
>>  _  _
>> |  A server  |  ___DTMF Send_=> | B server   |
>> |_|  <=--- Responce -   |_|
>>
>> *B server code is below:-*
>> At B server call come to 201 extension which is mention here..
>>
>> exten => _20[1-6],1,Answer()
>> same => n,Ringing()
>> same => n,wait(2)
>> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
>> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
>> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
>> $[${EXTEN}=205] ||
>> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
>> same => n,Hangup()
>>
>> Now I can send the DTMF from A to B. But How I will get the responce at
>> server A. I checked all the channels variable but they didn't reply status
>> of B server channel. All information I will get of server A. Main problem
>> is that control reach to AGI and then I don't have any rights to do any
>> update or modification on AGI. So if I can work on request and responce
>> then it will be the last solution as per my knowledge.
>>
>> Is this possible with the dialplan or I am just westing time?
>>
>>
>>
>> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger wrote:
>>
>>> On 11-12-28 03:25 AM, virendra bhati wrote:
>>>
 Hi list,

 Is there any way in asterisk by which I make a call from server and then
 dialplan(IVR system) gets DTMF from it. I mean to say that automatically
 DTMF is sended by channels as per user defined,

 I read there is an application sendDTMF but I don't know how we can
 used it?

 like A script make the call by using localdail, .call file or any
 method.
 And after landing the call we send dtmf to IVR system automatically as
 per
 my script..


 *extensions.conf:-*


 exten =>  1234,1,Answer()
  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
 rHelp,1,,10)
  same =>  n,NoOp(${value})
  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
  same =>  n,Hangup()

 exten=>  support,1,Answer()
  same =>  n,NoOp(you are at support section)
  same =>  n,Hangup()

 exten=>  help,1,Answer()
  same =>  n,NoOp(you are at help section)
  same =>  n,Hangup()

  We have DTMF based tests for the testsuite[1] that you could use.
>>>
>>> [1] 
>>> http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
>>> --
>>> Paul Belanger
>>> Digium, Inc. | Software Developer
>>> twitter: pabelanger | IRC: pabelanger (Freenode)
>>> Check us out at: http://digium.com & http://asterisk.org
>>>
>>>
>>> --
>>> __**__**
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>  http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>  
>>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>>
>>
>>
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>

Re: [asterisk-users] DTMF Testing software to test IVR system

[Sammy Govind]

Easy, use Read() to capture the incoming DTMF from Server-B

Server-A <> Server-B
Initiate-Call -> AnswerCall()
SendDTMF(5)--> Read()
Read()<-SendDTMF(4)
SendDTMF(3)--> Read()
Read()<-SendDTMF(2)
SendDTMF(1)--> Read()


Put proper GOTOIFs after reads if you like.

--
Regards,
Sammy

On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati  wrote:

> I originate calls from .call file and 1 channel I have at A server A and
> another channel at B server.
>
> *A server code is below:-*
>
> exten => 43689956,1,Answer()
> same => n,Wait(5)
> same => n,SendDTMF(1)
> same => n,NoOp(==   ${CHANNEL(state)}==> state)
> same => n,wait(2)
> same => n,SendDTMF(123456789012345#)
> same => n,NoOp(==   ${CHANNEL(state)}==> state)
> same => n,Hangup()
>
>  _  _
> |  A server  |  ___DTMF Send_=> | B server   |
> |_|  <=--- Responce -   |_|
>
> *B server code is below:-*
> At B server call come to 201 extension which is mention here..
>
> exten => _20[1-6],1,Answer()
> same => n,Ringing()
> same => n,wait(2)
> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
> $[${EXTEN}=205] ||
> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
> same => n,Hangup()
>
> Now I can send the DTMF from A to B. But How I will get the responce at
> server A. I checked all the channels variable but they didn't reply status
> of B server channel. All information I will get of server A. Main problem
> is that control reach to AGI and then I don't have any rights to do any
> update or modification on AGI. So if I can work on request and responce
> then it will be the last solution as per my knowledge.
>
> Is this possible with the dialplan or I am just westing time?
>
>
>
> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger wrote:
>
>> On 11-12-28 03:25 AM, virendra bhati wrote:
>>
>>> Hi list,
>>>
>>> Is there any way in asterisk by which I make a call from server and then
>>> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
>>> DTMF is sended by channels as per user defined,
>>>
>>> I read there is an application sendDTMF but I don't know how we can used
>>> it?
>>>
>>> like A script make the call by using localdail, .call file or any method.
>>> And after landing the call we send dtmf to IVR system automatically as
>>> per
>>> my script..
>>>
>>>
>>> *extensions.conf:-*
>>>
>>>
>>> exten =>  1234,1,Answer()
>>>  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
>>> rHelp,1,,10)
>>>  same =>  n,NoOp(${value})
>>>  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
>>>  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
>>>  same =>  n,Hangup()
>>>
>>> exten=>  support,1,Answer()
>>>  same =>  n,NoOp(you are at support section)
>>>  same =>  n,Hangup()
>>>
>>> exten=>  help,1,Answer()
>>>  same =>  n,NoOp(you are at help section)
>>>  same =>  n,Hangup()
>>>
>>>  We have DTMF based tests for the testsuite[1] that you could use.
>>
>> [1] 
>> http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
>> --
>> Paul Belanger
>> Digium, Inc. | Software Developer
>> twitter: pabelanger | IRC: pabelanger (Freenode)
>> Check us out at: http://digium.com & http://asterisk.org
>>
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>  http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>  
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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asterisk-user

Re: [asterisk-users] DTMF Testing software to test IVR system

[virendra bhati]

I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.

*A server code is below:-*

exten => 43689956,1,Answer()
same => n,Wait(5)
same => n,SendDTMF(1)
same => n,NoOp(==   ${CHANNEL(state)}==> state)
same => n,wait(2)
same => n,SendDTMF(123456789012345#)
same => n,NoOp(==   ${CHANNEL(state)}==> state)
same => n,Hangup()

 _  _
|  A server  |  ___DTMF Send_=> | B server   |
|_|  <=--- Responce -   |_|

*B server code is below:-*
At B server call come to 201 extension which is mention here..

exten => _20[1-6],1,Answer()
same => n,Ringing()
same => n,wait(2)
same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
$[${EXTEN}=205] ||
$[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
same => n,Hangup()

Now I can send the DTMF from A to B. But How I will get the responce at
server A. I checked all the channels variable but they didn't reply status
of B server channel. All information I will get of server A. Main problem
is that control reach to AGI and then I don't have any rights to do any
update or modification on AGI. So if I can work on request and responce
then it will be the last solution as per my knowledge.

Is this possible with the dialplan or I am just westing time?


On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger wrote:

> On 11-12-28 03:25 AM, virendra bhati wrote:
>
>> Hi list,
>>
>> Is there any way in asterisk by which I make a call from server and then
>> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
>> DTMF is sended by channels as per user defined,
>>
>> I read there is an application sendDTMF but I don't know how we can used
>> it?
>>
>> like A script make the call by using localdail, .call file or any method.
>> And after landing the call we send dtmf to IVR system automatically as per
>> my script..
>>
>>
>> *extensions.conf:-*
>>
>>
>> exten =>  1234,1,Answer()
>>  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
>> rHelp,1,,10)
>>  same =>  n,NoOp(${value})
>>  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
>>  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
>>  same =>  n,Hangup()
>>
>> exten=>  support,1,Answer()
>>  same =>  n,NoOp(you are at support section)
>>  same =>  n,Hangup()
>>
>> exten=>  help,1,Answer()
>>  same =>  n,NoOp(you are at help section)
>>  same =>  n,Hangup()
>>
>>  We have DTMF based tests for the testsuite[1] that you could use.
>
> [1] 
> http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>  http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>



-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
--
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Re: [asterisk-users] DTMF Testing software to test IVR system

[Paul Belanger]

On 11-12-28 03:25 AM, virendra bhati wrote:

Hi list,

Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,

I read there is an application sendDTMF but I don't know how we can used it?

like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..


*extensions.conf:-*

exten =>  1234,1,Answer()
  same =>  n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
  same =>  n,NoOp(${value})
  same =>  n,ExecIf($[${value}=1]?Goto(suppot,1))
  same =>  n,ExecIf($[${value}=2]?Goto(help,1))
  same =>  n,Hangup()

exten=>  support,1,Answer()
  same =>  n,NoOp(you are at support section)
  same =>  n,Hangup()

exten=>  help,1,Answer()
  same =>  n,NoOp(you are at help section)
  same =>  n,Hangup()


We have DTMF based tests for the testsuite[1] that you could use.

[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

--
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Re: [asterisk-users] DTMF Testing software to test IVR system

[Satish Barot]

What I understand from your reply is, you also like to have multiple Read()
in 'support' and 'help' extensions as well.

In that case you can have something like this in [senddtmf]
exten => s,1,Noop(# TEST:IVR ##)

; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same => n,Wait(10)
same => n,SendDTMF(1)
;- Wait for message in second Read --;
same => n,Wait(5)
same => n,SendDTMF(2)
;- Wait for message in third Read --;
same => n,Wait(15)
same => n,SendDTMF(1)
...
...
same => n,Wait(10)
same => n,SendDTMF(3)


Hope this helps you,

--SATISH BAROT


On Wed, Dec 28, 2011 at 3:02 PM, virendra bhati  wrote:

> Hi Satish,
>
> Thank you Satish. I did the same before your e-mail i saw. But i got
> another issue in such case.
> DTMF is passed to that channels but in case I will make the complete IVR
> system for calling server end. and which become so complected to do it.
>
> Is there any alternate way by which I get the response and send DTMF only.
> So that complete IVR flow willn't be required to implement at originator
> server.
>
>
> On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot 
> wrote:
>
>> Create a callfile with local channel and once first call leg is answered,
>> use wait() and senddtmf() application on second call leg.
>>
>>
>> CALLFILE sample:
>>
>> Channel: LOCAL/1234\@test_ivr
>> Context: senddtmf
>> Extension: s
>> Priority: 1
>>
>>
>> Extensions.conf sample:
>>
>> ;-- FIRST LEG CALL --;
>> [test_ivr]
>>
>> exten => 1234,1,Answer()
>> same => n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
>> same => n,NoOp(${value})
>> same => n,ExecIf($[${value}=1]?Goto(suppot,1))
>> same => n,ExecIf($[${value}=2]?Goto(help,1))
>> same => n,Hangup()
>>
>> exten=> support,1,Answer()
>> same => n,NoOp(you are at support section)
>> same => n,Hangup()
>>
>> exten=> help,1,Answer()
>> same => n,NoOp(you are at help section)
>> same => n,Hangup()
>>
>> ;--SECOND LEG CALL --;
>> [senddtmf]
>> exten => s,1,Noop(# TEST:IVR ##)
>>
>> ; We should wait atleast 'n' of seconds. Where n is length of IVR file in
>> seconds.
>> same => n,Wait(10)
>> same => n,SendDTMF(1)
>>
>>
>>
>>
>> --SATISH BAROT
>>
>> On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati wrote:
>>
>>> Hi list,
>>>
>>> Is there any way in asterisk by which I make a call from server and then
>>> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
>>> DTMF is sended by channels as per user defined,
>>>
>>> I read there is an application sendDTMF but I don't know how we can used
>>> it?
>>>
>>> like A script make the call by using localdail, .call file or any
>>> method. And after landing the call we send dtmf to IVR system automatically
>>> as per my script..
>>>
>>>
>>> *extensions.conf:-*
>>>
>>> exten => 1234,1,Answer()
>>>  same =>
>>> n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
>>>  same => n,NoOp(${value})
>>>  same => n,ExecIf($[${value}=1]?Goto(suppot,1))
>>>  same => n,ExecIf($[${value}=2]?Goto(help,1))
>>>  same => n,Hangup()
>>>
>>> exten=> support,1,Answer()
>>>  same => n,NoOp(you are at support section)
>>>  same => n,Hangup()
>>>
>>> exten=> help,1,Answer()
>>>  same => n,NoOp(you are at help section)
>>>  same => n,Hangup()
>>>
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
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New

Re: [asterisk-users] DTMF Testing software to test IVR system

[Sammy Govind]

Hi,
You can use combination of SendDTMF() and wait() in such a way that you
traverse through the IVR tree just as Satish mentioned.

SendDTMF(1)
Wait(3)
SendDTMF(2)
Wait(2)
SendDTMF(5678123490)

 See also:
*WaitForNoise()* ,  WaitForSilence(), AMD()

Regards,
Sammy.

On Wed, Dec 28, 2011 at 2:32 PM, virendra bhati  wrote:

> Hi Satish,
>
> Thank you Satish. I did the same before your e-mail i saw. But i got
> another issue in such case.
> DTMF is passed to that channels but in case I will make the complete IVR
> system for calling server end. and which become so complected to do it.
>
> Is there any alternate way by which I get the response and send DTMF only.
> So that complete IVR flow willn't be required to implement at originator
> server.
>
>
> On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot 
> wrote:
>
>> Create a callfile with local channel and once first call leg is answered,
>> use wait() and senddtmf() application on second call leg.
>>
>>
>> CALLFILE sample:
>>
>> Channel: LOCAL/1234\@test_ivr
>> Context: senddtmf
>> Extension: s
>> Priority: 1
>>
>>
>> Extensions.conf sample:
>>
>> ;-- FIRST LEG CALL --;
>> [test_ivr]
>>
>> exten => 1234,1,Answer()
>> same => n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
>> same => n,NoOp(${value})
>> same => n,ExecIf($[${value}=1]?Goto(suppot,1))
>> same => n,ExecIf($[${value}=2]?Goto(help,1))
>> same => n,Hangup()
>>
>> exten=> support,1,Answer()
>> same => n,NoOp(you are at support section)
>> same => n,Hangup()
>>
>> exten=> help,1,Answer()
>> same => n,NoOp(you are at help section)
>> same => n,Hangup()
>>
>> ;--SECOND LEG CALL --;
>> [senddtmf]
>> exten => s,1,Noop(# TEST:IVR ##)
>>
>> ; We should wait atleast 'n' of seconds. Where n is length of IVR file in
>> seconds.
>> same => n,Wait(10)
>> same => n,SendDTMF(1)
>>
>>
>>
>>
>> --SATISH BAROT
>>
>> On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati wrote:
>>
>>> Hi list,
>>>
>>> Is there any way in asterisk by which I make a call from server and then
>>> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
>>> DTMF is sended by channels as per user defined,
>>>
>>> I read there is an application sendDTMF but I don't know how we can used
>>> it?
>>>
>>> like A script make the call by using localdail, .call file or any
>>> method. And after landing the call we send dtmf to IVR system automatically
>>> as per my script..
>>>
>>>
>>> *extensions.conf:-*
>>>
>>> exten => 1234,1,Answer()
>>>  same =>
>>> n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
>>>  same => n,NoOp(${value})
>>>  same => n,ExecIf($[${value}=1]?Goto(suppot,1))
>>>  same => n,ExecIf($[${value}=2]?Goto(help,1))
>>>  same => n,Hangup()
>>>
>>> exten=> support,1,Answer()
>>>  same => n,NoOp(you are at support section)
>>>  same => n,Hangup()
>>>
>>> exten=> help,1,Answer()
>>>  same => n,NoOp(you are at help section)
>>>  same => n,Hangup()
>>>
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] DTMF Testing software to test IVR system

[virendra bhati]

Hi Satish,

Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.

Is there any alternate way by which I get the response and send DTMF only.
So that complete IVR flow willn't be required to implement at originator
server.

On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot wrote:

> Create a callfile with local channel and once first call leg is answered,
> use wait() and senddtmf() application on second call leg.
>
>
> CALLFILE sample:
>
> Channel: LOCAL/1234\@test_ivr
> Context: senddtmf
> Extension: s
> Priority: 1
>
>
> Extensions.conf sample:
>
> ;-- FIRST LEG CALL --;
> [test_ivr]
>
> exten => 1234,1,Answer()
> same => n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
> same => n,NoOp(${value})
> same => n,ExecIf($[${value}=1]?Goto(suppot,1))
> same => n,ExecIf($[${value}=2]?Goto(help,1))
> same => n,Hangup()
>
> exten=> support,1,Answer()
> same => n,NoOp(you are at support section)
> same => n,Hangup()
>
> exten=> help,1,Answer()
> same => n,NoOp(you are at help section)
> same => n,Hangup()
>
> ;--SECOND LEG CALL --;
> [senddtmf]
> exten => s,1,Noop(# TEST:IVR ##)
>
> ; We should wait atleast 'n' of seconds. Where n is length of IVR file in
> seconds.
> same => n,Wait(10)
> same => n,SendDTMF(1)
>
>
>
>
> --SATISH BAROT
>
> On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati wrote:
>
>> Hi list,
>>
>> Is there any way in asterisk by which I make a call from server and then
>> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
>> DTMF is sended by channels as per user defined,
>>
>> I read there is an application sendDTMF but I don't know how we can used
>> it?
>>
>> like A script make the call by using localdail, .call file or any method.
>> And after landing the call we send dtmf to IVR system automatically as per
>> my script..
>>
>>
>> *extensions.conf:-*
>>
>> exten => 1234,1,Answer()
>>  same =>
>> n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
>>  same => n,NoOp(${value})
>>  same => n,ExecIf($[${value}=1]?Goto(suppot,1))
>>  same => n,ExecIf($[${value}=2]?Goto(help,1))
>>  same => n,Hangup()
>>
>> exten=> support,1,Answer()
>>  same => n,NoOp(you are at support section)
>>  same => n,Hangup()
>>
>> exten=> help,1,Answer()
>>  same => n,NoOp(you are at help section)
>>  same => n,Hangup()
>>
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] DTMF Testing software to test IVR system

[Satish Barot]

Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.


CALLFILE sample:

Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1


Extensions.conf sample:

;-- FIRST LEG CALL --;
[test_ivr]
exten => 1234,1,Answer()
same => n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same => n,NoOp(${value})
same => n,ExecIf($[${value}=1]?Goto(suppot,1))
same => n,ExecIf($[${value}=2]?Goto(help,1))
same => n,Hangup()

exten=> support,1,Answer()
same => n,NoOp(you are at support section)
same => n,Hangup()

exten=> help,1,Answer()
same => n,NoOp(you are at help section)
same => n,Hangup()

;--SECOND LEG CALL --;
[senddtmf]
exten => s,1,Noop(# TEST:IVR ##)

; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same => n,Wait(10)
same => n,SendDTMF(1)




--SATISH BAROT

On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati  wrote:

> Hi list,
>
> Is there any way in asterisk by which I make a call from server and then
> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
> DTMF is sended by channels as per user defined,
>
> I read there is an application sendDTMF but I don't know how we can used
> it?
>
> like A script make the call by using localdail, .call file or any method.
> And after landing the call we send dtmf to IVR system automatically as per
> my script..
>
>
> *extensions.conf:-*
>
> exten => 1234,1,Answer()
>  same =>
> n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
>  same => n,NoOp(${value})
>  same => n,ExecIf($[${value}=1]?Goto(suppot,1))
>  same => n,ExecIf($[${value}=2]?Goto(help,1))
>  same => n,Hangup()
>
> exten=> support,1,Answer()
>  same => n,NoOp(you are at support section)
>  same => n,Hangup()
>
> exten=> help,1,Answer()
>  same => n,NoOp(you are at help section)
>  same => n,Hangup()
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks [WORKING]

[JR Richardson]

> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in
> routing calls to upstream carrier via SIP trunks out.  I spent a lot of time
> in the lab testing 1.8 which included heavily testing DTMF with no issues
> that came up.  It all just seemed to work fine.  But then again you can’t
> reproduce every real work scenario in the lab.
>
>
>
> I’m using rfc2833 inbound and outbound for the new 1.8 call servers.  Here
> is a quick diagram of what is working and what is not:
>
>
>
> Not working:
>
> Customer IP PBX> ast 1.8 rfc2833>
>
>
> Customer PRI> trunk>< call server ast 1.8 rfc2833>
>
>
> I can see DTMF RTP events pass through call server to carrier but no
> response, nothing, nada, zip.
>
>
>
> Working:
>
> Customer SIP Phone>< call server
> ast 1.8 rfc2833>
>
>
> Customer SIP Phone>< call server
> ast 1.2 rfc2833>
>
>
> Customer IP PBX>< call server
> ast 1.2 rfc2833>
>
>
> Customer PRI>< call
> server sip trunk>
>
>
> I can see DTMF RTP events pass through to carrier, RTP stream looks the same
> as the 1.8 server with reliable responses.
>
>
>
> On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active on
> peer and global settings:
>
> relaxdtmf=yes
>
> rfc2833compensate=yes
>
> dtmfmode=rfc2833
>
>
>
> Now it quickly appears like a problem between the customer PBX and Customer
> PRI with the SIP trunks to the ast 1.4 servers but it all worked fine before
> with the 1.2 call servers.  After the upgrade of the call servers to 1.8
> DTMF is not recognized by the carrier on calls from the customer IP PBX or
> PRI but is fine with the SIP phones directly registered to the ast 1.4
> servers.
>
>
>
> I found the bug issues with the SRCC change/update issues with DTMF events.
> It looks like 1.8.6.0 implemented the ‘update’ and as I read it, should have
> fixed the issue with the changing SRCC effecting DTMF.  But this may not be
> the case.
>
>
>
> Specifically, how would I debug RTP/DTMF on the new ast 1.8 server and see
> if the SRCC is changing between my scenarios described above.  Am I on the
> right track or is there something else I should be looking at?
>
I added [96] in */main/rtp.c of the ast 1.4 servers then recompiled.
[96] = {0, AST_RTP_DTMF},
[97] = {1, AST_FORMAT_ILBC},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
This seemed to allow flow through of the DTMF up to the new 1.8 call
servers and on to the carrier.  I don't know why this seemed to fix
the issue, I'm not 100% convinced it really did.  I reverted the
change and could not reproduce the issue, so I also suspect the
upstream carrier started working or may have changed something
coincidentaly.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses

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Re: [asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks

[Jared Geiger]

I had similar problems with 1.8.6 and polycom phones intermittently having
DTMF issues. I updated to 1.8.7 and things cleared up. I went through the
release notes at the time, but don't recall which commit made me decide to
give it a try.

Rgds,
Jared

On Wed, Nov 9, 2011 at 7:03 PM, JR Richardson wrote:

>  Hi All,
>
> ** **
>
> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks
> in routing calls to upstream carrier via SIP trunks out.  I spent a lot of
> time in the lab testing 1.8 which included heavily testing DTMF with no
> issues that came up.  It all just seemed to work fine.  But then again you
> can’t reproduce every real work scenario in the lab.
>
> ** **
>
> I’m using rfc2833 inbound and outbound for the new 1.8 call servers.  Here
> is a quick diagram of what is working and what is not:
>
> ** **
>
> Not working:
>
> Customer IP PBX> server ast 1.8 rfc2833>
> ** **
>
> Customer PRI> trunk>< call server ast 1.8 rfc2833>
> ** **
>
> I can see DTMF RTP events pass through call server to carrier but no
> response, nothing, nada, zip.
>
> ** **
>
> Working:
>
> Customer SIP Phone>< call server
> ast 1.8 rfc2833>
> ** **
>
> Customer SIP Phone>< call server
> ast 1.2 rfc2833>
> ** **
>
> Customer IP PBX>< call
> server ast 1.2 rfc2833>
> ** **
>
> Customer PRI>< call
> server sip trunk>
> ** **
>
> I can see DTMF RTP events pass through to carrier, RTP stream looks the
> same as the 1.8 server with reliable responses.
>
> ** **
>
> On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active
> on peer and global settings:
>
> relaxdtmf=yes
>
> rfc2833compensate=yes
>
> dtmfmode=rfc2833
>
> ** **
>
> Now it quickly appears like a problem between the customer PBX and
> Customer PRI with the SIP trunks to the ast 1.4 servers but it all worked
> fine before with the 1.2 call servers.  After the upgrade of the call
> servers to 1.8 DTMF is not recognized by the carrier on calls from the
> customer IP PBX or PRI but is fine with the SIP phones directly registered
> to the ast 1.4 servers.
>
> ** **
>
> I found the bug issues with the SRCC change/update issues with DTMF
> events.  It looks like 1.8.6.0 implemented the ‘update’ and as I read it,
> should have fixed the issue with the changing SRCC effecting DTMF.  But
> this may not be the case.
>
> ** **
>
> Specifically, how would I debug RTP/DTMF on the new ast 1.8 server and see
> if the SRCC is changing between my scenarios described above.  Am I on the
> right track or is there something else I should be looking at?
>
> ** **
>
> Thanks.
>
>
> JR
>
> --
> _
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Re: [asterisk-users] DTMF fun

[Benny Amorsen]

Tom Browning  writes:

> My question is this:  Is Asterisk simply relaying the client's DTMF
> signalling untouched or do I need to look at Asterisk more
> closely and turn some knobs.

I would recommend that you grab some wireshark traces before and after
the DTMF traverses Asterisk. It should be fairly easy to verify whether
Asterisk changes the length or number of DTMF messages.


/Benny


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Re: [asterisk-users] DTMF problem

[Daniel Tryba]

On Sun, Sep 18, 2011 at 07:51:43PM -0400, Zeeshan A Zakaria wrote:
> This DTMF problem has always been there and there is no real solution
> for it, other than using those expensive PBX systems like that from
> Avaya, Cisco, etc. This problem happens when you are sending inband
> DTMF tones. Via softphone you are sending out-of-band DTMF which is
> basically SIP messages.

You can emulate this feature from the Expensive PBX system by setting:
relaxdtmf=yes
in the case of SIP, option may vary with Techology.

-- 

   Daniel Tryba

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Re: [asterisk-users] DTMF problem

[Olle E. Johansson]

19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria:

> This DTMF problem has always been there and there is no real solution for it, 
> other than using those expensive PBX systems like that from Avaya, Cisco, 
> etc. This problem happens when you are sending inband DTMF tones. Via 
> softphone you are sending out-of-band DTMF which is basically SIP messages.

Just to correct the latest part of your statement:

The default way to send DTMF in SIP calls is using DTMF as a codec called 
telephony-event in the RTP stream. This sends DTMF as events. Most hard and 
soft phones support this - usually called RFC2833 DTMF mode. Asterisk supports 
it by default. 

Sending DTMF in the audio usually gets messy when using an IP network. 
Especially if you use codecs that compress the audio. I do recommend you to use 
RFC2833. We have built very large IVR services and have no issues with DTMF 
being received in Asterisk so it's doable.

There are other issues with Asterisk DTMF, but that's another issue :-)

/O




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Re: [asterisk-users] DTMF problem

[Sam Govind]

Hey there,

I don't think that its DTMF mode issue ! OP say pressing 9 asterisk ignores
while pressing 6 is OK. Using expensive PBX solutions should be never be the
first priority.

So I'd a similar experience in some asterisk version when I used to enter 2
asterisk always took 3-4 seconds to do anything wheras all other DTMF digits
were working as quickly as DTMF entered.
Since pressing 6 key works fine means this could be more possibly issue with
handset.
Also there is an option in background application "m" where background will
only accept the DTMF whose extens are created in the same context..so if
you've something like this in your dialplan

[test-BKGRND]
exten => s,1,Background(som-sound-file,m)
exten => s,n,Waitexten(20)

exten => 1,1,NOOP(User presssed 1)
exten => 3,1,NOOP(User presssed 1)
exten => 5,1,NOOP(User presssed 1)

Background will act as only recognizing DTMF 1,3,and 5.

even if its DTmfmode issue ...you can one more trick to fix this as well..
use application sipdtmfmode(inband|rfc2833|info)  if call is coming from a
particular caller/UA.

I hope this could be of some help.

Regards,

- Sammy

On Mon, Sep 19, 2011 at 4:51 AM, Zeeshan A Zakaria wrote:

> This DTMF problem has always been there and there is no real solution for
> it, other than using those expensive PBX systems like that from Avaya,
> Cisco, etc. This problem happens when you are sending inband DTMF tones. Via
> softphone you are sending out-of-band DTMF which is basically SIP messages.
>
> --
> Zeeshan A Zakaria
>
> IT Consultant
> www.zeeshanz.com
> 855-ZEESHAN
>
> "asterisk asterisk"  wrote:
>
> >From time to time, I have a DTMF problem. The asterisk cannot recognize
> >my
> >handset key pressed if I press 9 to start with. However, if I press
> >with 6,
> >it is ok.
> >
> >On the other hand, if DMTF key is generated from softphone, it is ok.
> >
> >My dialplan is as follow
> >
> >exten => 1002,1,Answer
> >exten => 1002,n,Wait(2)
> >exten => 1002,n,Background(thank-you-for-calling)
> >exten => 1002,n,Background(vm-enter-num-to-call)
> >exten => 1002,n,WaitExten()
> >exten => 1002,n,Hangup
> >exten => i,1,Background(pbx-invalid)
> >exten => i,2,Goto(1002,1)
> >exten => t,1,Background(vm-goodbye)
> >exten => t,2,Hangup
> >
> >Thanks for the help in advance.
> >--
> >_
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> >
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Sent from my Android phone with K-9 Mail. Please excuse my brevity.
>
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Re: [asterisk-users] DTMF problem

[Zeeshan A Zakaria]

This DTMF problem has always been there and there is no real solution for it, 
other than using those expensive PBX systems like that from Avaya, Cisco, etc. 
This problem happens when you are sending inband DTMF tones. Via softphone you 
are sending out-of-band DTMF which is basically SIP messages.

--
Zeeshan A Zakaria

IT Consultant
www.zeeshanz.com
855-ZEESHAN

"asterisk asterisk"  wrote:

>From time to time, I have a DTMF problem. The asterisk cannot recognize
>my
>handset key pressed if I press 9 to start with. However, if I press
>with 6,
>it is ok.
>
>On the other hand, if DMTF key is generated from softphone, it is ok.
>
>My dialplan is as follow
>
>exten => 1002,1,Answer
>exten => 1002,n,Wait(2)
>exten => 1002,n,Background(thank-you-for-calling)
>exten => 1002,n,Background(vm-enter-num-to-call)
>exten => 1002,n,WaitExten()
>exten => 1002,n,Hangup
>exten => i,1,Background(pbx-invalid)
>exten => i,2,Goto(1002,1)
>exten => t,1,Background(vm-goodbye)
>exten => t,2,Hangup
>
>Thanks for the help in advance.
>--
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>   http://www.asterisk.org/hello
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-- 
Sent from my Android phone with K-9 Mail. Please excuse my brevity.

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Re: [asterisk-users] DTMF games with Asterisk

[C. Savinovich]

You have plenty of ways to do this.  You can use the room number + user number
to get the conference number. You can use the channel ids to keep a table of
conference members and their statuses.
 
C. Savinovich 
 
 


On September 7, 2011 at 9:15 AM Danny Nicholas  wrote:


> 
> It seems to me that you are overworking AMI to do what could be done with
> AGI.  You could use an AGI to poll Konference and return a dialplan variable
> with the file to use in Playback/Background or even MOH.
>  
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
> Sent: Wednesday, September 07, 2011 6:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion; onewaytoconnect
> Subject: Re: [asterisk-users] DTMF games with Asterisk
>  
> 
> Hi Amit,
> 
> My scenario is that, If 3 conference is running in Asterisk then I will play a
> sound file with the help of Asterisk AMI then I will get DTMF from all the
> users. the same things will be done any all the Konference and all conference
> will be play different files.
> 
> If you have any alternate suggestion the please help me
> 
> On Wed, Sep 7, 2011 at 5:00 PM, amit anand  [mailto:onewaytoconn...@gmail.com] > wrote:
> Hi
> 
> This can happen you can create more than  1 AMI connection.
> 
> if you need better on access control then you can create new user in
> manager.conf with set of privileges that you can offer to each of them
> 
> 
> 
> On Wed, Sep 7, 2011 at 15:59, virendra bhati  [mailto:virbh...@gmail.com] > wrote:
> 
> > 
> > 
> > 
> > Hi list,
> >  
> > I want to know that will it be possible that more then 1 AMI is connected
> > from single Linux machine with different name ?
> > 
> > As we know that default 1st AMI connection will come with 127.0.0.1 and root
> > information.
> > 
> > My requirement is that I want to handling events for more then one
> > Konference. So I required more then 1 AMI connection might be 1 connection
> > for 1 konference. Because I will play some IVR files to get DTMF and on this
> > DTMF i will check the correct DTMF. So that I will get the right user with
> > correct input.
> > 
> > So please guide me.
> > 
> > --
> > 
> > 
> > 
> > 
> > -
> > Thanks and regards
> > 
> >  Virendra Bhati
> > +91-9172341457 [tel:%2B91-9172341457]
> > Software Engineer
> >  
> >  
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >               http://www.asterisk.org/hello
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > 

> 
> 
> --
> 
>  
> 
> Amit Anand
> 
>  
> 
>  
> 
> 
> +91 9818559898 
> 
>  
>  
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>               http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> --
> 
> 
> 
> 
> -
> Thanks and regards
> 
>  Virendra Bhati
> +91-9172341457
> Software Engineer
>  
> 
Christian Savinovich
Telecom & Telephony Consulting
646.982.3572
c.savinov...@itntelecom.com--
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Re: [asterisk-users] DTMF games with Asterisk

[Danny Nicholas]

It seems to me that you are overworking AMI to do what could be done with
AGI.  You could use an AGI to poll Konference and return a dialplan variable
with the file to use in Playback/Background or even MOH.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, September 07, 2011 6:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; onewaytoconnect
Subject: Re: [asterisk-users] DTMF games with Asterisk

 

Hi Amit,

My scenario is that, If 3 conference is running in Asterisk then I will play
a sound file with the help of Asterisk AMI then I will get DTMF from all the
users. the same things will be done any all the Konference and all
conference will be play different files. 

If you have any alternate suggestion the please help me 

On Wed, Sep 7, 2011 at 5:00 PM, amit anand 
wrote:

Hi

This can happen you can create more than  1 AMI connection.

if you need better on access control then you can create new user in
manager.conf with set of privileges that you can offer to each of them

On Wed, Sep 7, 2011 at 15:59, virendra bhati  wrote:

Hi list,
 
I want to know that will it be possible that more then 1 AMI is connected
from single Linux machine with different name ?

As we know that default 1st AMI connection will come with 127.0.0.1 and root
information.

My requirement is that I want to handling events for more then one
Konference. So I required more then 1 AMI connection might be 1 connection
for 1 konference. Because I will play some IVR files to get DTMF and on this
DTMF i will check the correct DTMF. So that I will get the right user with
correct input.

So please guide me. 

-- 




-
Thanks and regards

 Virendra Bhati
+91-9172341457  
Software Engineer

 

 

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-- 

 

Amit Anand

 


 


+91 9818559898

 

 


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-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer

 

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Re: [asterisk-users] DTMF games with Asterisk

[virendra bhati]

Hi Amit,

My scenario is that, If 3 conference is running in Asterisk then I will play
a sound file with the help of Asterisk AMI then I will get DTMF from all the
users. the same things will be done any all the Konference and all
conference will be play different files.

If you have any alternate suggestion the please help me

On Wed, Sep 7, 2011 at 5:00 PM, amit anand wrote:

> Hi
>
> This can happen you can create more than  1 AMI connection.
>
> if you need better on access control then you can create new user in
> manager.conf with set of privileges that you can offer to each of them
>
> On Wed, Sep 7, 2011 at 15:59, virendra bhati  wrote:
>
>> Hi list,
>>
>> I want to know that will it be possible that more then 1 AMI is connected
>> from single Linux machine with different name ?
>>
>> As we know that default 1st AMI connection will come with 127.0.0.1 and
>> root information.
>>
>> My requirement is that I want to handling events for more then one
>> Konference. So I required more then 1 AMI connection might be 1 connection
>> for 1 konference. Because I will play some IVR files to get DTMF and on this
>> DTMF i will check the correct DTMF. So that I will get the right user with
>> correct input.
>>
>> So please guide me.
>>
>> --
>>
>>
>>
>> -
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9172341457
>> Software Engineer
>>
>>
>> --
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Amit Anand
>
>
> +91 9818559898
>
>
>
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-- 



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Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] DTMF games with Asterisk

[amit anand]

Hi

This can happen you can create more than  1 AMI connection.

if you need better on access control then you can create new user in
manager.conf with set of privileges that you can offer to each of them

On Wed, Sep 7, 2011 at 15:59, virendra bhati  wrote:

> Hi list,
>
> I want to know that will it be possible that more then 1 AMI is connected
> from single Linux machine with different name ?
>
> As we know that default 1st AMI connection will come with 127.0.0.1 and
> root information.
>
> My requirement is that I want to handling events for more then one
> Konference. So I required more then 1 AMI connection might be 1 connection
> for 1 konference. Because I will play some IVR files to get DTMF and on this
> DTMF i will check the correct DTMF. So that I will get the right user with
> correct input.
>
> So please guide me.
>
> --
>
>
>
> -
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Amit Anand


+91 9818559898
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Re: [asterisk-users] DTMF issues still

[vmedina]

Latest firmware is  on the card

Sent from my android device.

-Original Message-
From: Jim Dickenson 
To: asterisk-users@lists.digium.com
Sent: Fri, 08 Jul 2011 5:59 PM
Subject: Re: [asterisk-users] DTMF issues still

I had a very strange problem with a Sangoma card that I had both Sangoma (about 
3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma 
tech to look at the problem it went away. I told the tech he did something and 
he said I alway verify the firmware on the card is updated and as it was not I 
updated it. That fixed the problem.


This system had worked before a dahdi update was applied.


Bottom line make sure you have the most current firmware for your card.

-- 

Jim Dickenson

mailto:dicken...@cfmc.com


CfMC

http://www.cfmc.com/




On Jul 8, 2011, at 2:49 PM, vmed...@apcn.net wrote:


I am still having major issues with dtmf recognition. My setup is Polycom end 
points. Tried this with different models, firmware and cfgs. Outbound calls are 
not going out reliably. Phones are set to rfc2833. I have had sangoma and 
elastix support look at it.. No better. Running asterisk 1.8.4. What am I 
missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on 
all lines no problem. Sangoma card is a a400 with echo cancel.


Sent from my android device.

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Re: [asterisk-users] DTMF issues still

[Jim Dickenson]

I had a very strange problem with a Sangoma card that I had both Sangoma (about 
3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma 
tech to look at the problem it went away. I told the tech he did something and 
he said I alway verify the firmware on the card is updated and as it was not I 
updated it. That fixed the problem.

This system had worked before a dahdi update was applied.

Bottom line make sure you have the most current firmware for your card.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jul 8, 2011, at 2:49 PM, vmed...@apcn.net wrote:

> I am still having major issues with dtmf recognition. My setup is Polycom end 
> points. Tried this with different models, firmware and cfgs. Outbound calls 
> are not going out reliably. Phones are set to rfc2833. I have had sangoma and 
> elastix support look at it.. No better. Running asterisk 1.8.4. What am I 
> missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on 
> all lines no problem. Sangoma card is a a400 with echo cancel.
> 
> 
> Sent from my android device.
> 
> --
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Re: [asterisk-users] DTMF begin ignored

[Marcelo Ellmann Clemente]

I'm not Asterisk expert but I had some DTMF problems in the recent past...

This message is called on main/channel.c everytime a DTMF is received, and, 
afaik, this is not an error, it Asterisk ignores the first milisenconds of the 
DTMF to distinguish between a real DTMF and any sound in the same frequency of 
that DTMF that may come in the call (and is not a real DTMF).

I was having some problems with the E1 card I was using and no DTMF mas 
detected at all. I've fixed it (using another card :p) and I've got a full 
system in production (with a quite heavy load of calls and DTMF inputs) and 
that DTMF begin ignored is quite normal during all key presses.

Can you show a piece of code of the extension/context which is reading this 
DTMF? Maybe that would help!

Good luck,
--- 
Marcelo Ellmann 
Freeddom Tecnologia e Serviços S/A
+55 11 52133200 Ramal 1016




- Original Message -
From: "vip killa" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, 21 June, 2011 2:07:37 PM
Subject: [asterisk-users] DTMF begin ignored


we've been getting complaints that DTMF is not working, i checked full log for 
a call that they claimed DTMF didnt work, I noticed this: 
DTMF begin '7' received 
DTMF begin ignored 
DTMF end '7' received 
DTMF end passthrough '7' 


why is the "DTMF begin ignored" called? 
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Re: [asterisk-users] dtmf Caller-id detection before first ring

[Ashik Ali]

We don't have polarity reversal before first ring I think so. Not confirmed.

 I able to see and hear dtmf and ring tones while playing recorded
wave file using audacity. As per the instruction given by Mr. Pezhman
Lai. I found somthing
while googling . The bug instructions as follows
https://issues.asterisk.org/view.php?id=9096&nbn=2.

I am using  8 port digium tdm card. So I decided edit
wctdm24xxp/base.c as per the instruction written in above issue.  As
of now I didn't touch asterisk chan_dahdi.c.

I modified base.c code as follows,

static void wctdm_dtmfcheck_fakepolarity(struct wctdm *wc, int card, u8 sample8)
{
u32 sample16;
struct fxo *const fxo = &(wc->mods[card].fxo);

/* only look for sound on the line if dtmf flag is on, it is an fxo
 * card and line is onhook */
if (!dtmf || !(wc->cardflag & (1 << card)) ||
(wc->modtype[card] != MOD_TYPE_FXO) || fxo->offhook) {
return;
}

/* don't look for noise if we're already processing it, or there is a
 * ringing tone */
if (!fxo->readcid && !fxo->wasringing  &&
wc->intcount > fxo->cidtimer + 400) {
sample16 = DAHDI_XLAW(sample8, wc->chans[card]);
if (sample16 > 2000 || sample16 < -2000) {
fxo->readcid = 1;
fxo->cidtimer = wc->intcount;
if (debug && ( card == 2 )) {
printk(KERN_DEBUG "DTMF CLIP on
<<<%i>>> <%X>\n",
   card + 1,sample16);
}
//  dahdi_qevent_lock(wc->chans[card],
//DAHDI_EVENT_POLARITY);
}
} else if (fxo->readcid && wc->intcount > fxo->cidtimer + 2000) {
/* reset flags if it's been a while */
fxo->cidtimer = wc->intcount;
fxo->readcid = 0;
}
}

After compilation of above, I just restarted dahdi and monitored
kernel message. I got following messages before receiving call


DTMF CLIP on <<<3>>> <68>
DTMF CLIP on <<<3>>> <84>
DTMF CLIP on <<<3>>> <58>
DTMF CLIP on <<<3>>> <78>
DTMF CLIP on <<<3>>> <60>
DTMF CLIP on <<<3>>> <48>
DTMF CLIP on <<<3>>> <60>
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on <<<3>>> <84>
DTMF CLIP on <<<3>>> <78>
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on <<<3>>> <68>
DTMF CLIP on <<<3>>> <68>
DTMF CLIP on <<<3>>> <70>
DTMF CLIP on <<<3>>> 
DTMF CLIP on <<<3>>> <70>

But some calls, I am getting long hexadecimall value  as follows,

DTMF CLIP on <<<3>>> <48>
DTMF CLIP on <<<3>>> 
RING on 1/3!
NO RING on 1/3!
DTMF CLIP on <<<3>>> <84>
DTMF CLIP on <<<3>>> <78>
RING on 1/3!
NO RING on 1/3!


Can u guide me on right Mr.Lali and Cohen.. ?


Thanks & Regards,
Ashik Ali


On Sun, May 29, 2011 at 12:13 PM, Tzafrir Cohen
 wrote:
> On Sat, May 28, 2011 at 02:34:36PM +0300, Ashik Ali wrote:
>> Hi dears,
>>
>> I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and
>> Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) .
>>
>> I am facing problem with detecting caller id before first ring.I
>> recorded the dahdi channel using dahdi_monitor command. Where I am
>> able to see and hear caller-id dtmf tones.
>
> Is there a polarity reversal before the caller ID string is sent?
>
> --
>               Tzafrir Cohen
> icq#16849755              jabber:tzafrir.co...@xorcom.com
> +972-50-7952406           mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] dtmf Caller-id detection before first ring

[Tzafrir Cohen]

On Sat, May 28, 2011 at 02:34:36PM +0300, Ashik Ali wrote:
> Hi dears,
> 
> I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and
> Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) .
> 
> I am facing problem with detecting caller id before first ring.I
> recorded the dahdi channel using dahdi_monitor command. Where I am
> able to see and hear caller-id dtmf tones.

Is there a polarity reversal before the caller ID string is sent?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] dtmf Caller-id detection before first ring

[Pezhman Lali]

you have to do these:

1-find suitable patch for your driver(wctdm.c) where the cidbeforering will
be defined.
2-modify the chan_dahdi.c in asterisk, change res to 4000 or higher
3-recompile your driver and asterisk
4-set  cidbeforering=1 and cidstart,  in the config of dahdi.
5-restart your machine.

for hearing the callerid before first ring use some sound editor for example
waveditor, to see what you hear.
best

On Sat, May 28, 2011 at 4:04 PM, Ashik Ali wrote:

> Hi dears,
>
> I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and
> Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) .
>
> I am facing problem with detecting caller id before first ring.I
> recorded the dahdi channel using dahdi_monitor command. Where I am
> able to see and hear caller-id dtmf tones.
>
> Pl tell me the procedure to upload recorded file if you needed.
> Something I want to dig it and make it work in asterisk.
>
> Thanks & Regards,
> Ashik
>
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-- 
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Re: [asterisk-users] DTMF not being sent ( RFC2833 )

[Jim Dickenson]

I had problems with a system I was trying to bring up using a couple older 
a104d cards we had lying around. Neither card would pass audio. I worked with 
one Sangoma tech for a couple hours while he tried various things. The second 
tech I worked with got on the system and updated the firmware for the cards. 
When I tried to show him the problem things worked. I said "you did something 
as this did not work an hour ago". He told me the first think he does when 
troubleshooting is to update the firmware to the current version. A lesson I 
have now learned. I do that with software but rarely remember to look for 
firmware updates. Take a look at wiki.sangoma.com and it lets you know current 
firmware versions as well as how to update if you are not running the current 
version.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Apr 25, 2011, at 4:41 PM, Edwin Lam wrote:

> i think i have similar problem after upgraded from 1.4.x to 1.6.2.17.
> (originally upgraded to 1.8.3.2 unfortunately there were other more
> pressing problems that forced me to downgraded it to 1.6.2.17)
> i have a wanpipe device with 2 channels uses PRI signalling to PSTN &
> the other 2 uses FXO signalling (connect to Rhino FXS channel bank).
> the PRI part works fine but the FXO channels are having DTMF digits
> skipped. i'm still trying to find out what's wrong with it.
> 
> On 4/23/11 8:48 AM, David wrote:
>> Hello,
>> I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple
>> problems with DTMF.
>> I have two machines, we'll call them asterisk and asterisk-pri. Asterisk 
>> does IVR
>> and asterisk-pri has a PRI card in it and connects to the PSTN. The two 
>> servers
>> communicate via SIP with RFC2833.
>> I setup logger.conf on both machines to display DTMF to the console. Both are
>> built from source.
>> Asterisk : spandsp, dahdi, asterisk.
>> Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe
>> I eliminated AGI, hard phones, network et al by setting up this extension :
>> exten => 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983
>> #))
>> in default.
>> The only other non default setting is in sip.conf I added a outboundproxy ( 
>> which
>> does NOT do RTP, only SIP ).
>> I called asterisk from my hard phone ( gxp2000 ) by dialing 22.
>> I see the console DTMF messages indicating the DTMF was sent or received. ( I
>> forgot to keep this output ).
>> I than watch the console DTMF output on asterisk-pri and it showed about 
>> half the
>> DTMFs. The pager that was called showed the DTMFs that appeared on the
>> asterisk-pri console.
>> So somewhere between the two machines, the DTMFs have disappeared. So I ran
>> TCPDump on asterisk and saw that close to half of the DTMF events were never 
>> sent.
>> tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap
>> I imported the file into wireshark on my local machine and confirmed that 
>> the dump
>> almost matches what I saw on asterisk-pri.
>> So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri.
>> I compared the packet scan to what I saw on asterisk-pri and noticed that 
>> between
>> 1 and 3 dtmfs were missing.
>> Problem 2 : Asterisk-pri loses some received DTMFs.
>> I also noticed that some of the DTMFs coming out of asterisk had the wrong 
>> Event
>> Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58
>> seconds ) but I only pressed the button for like 1/3 of a second.
>> What I do not understand is that I in my final test last night was using 
>> asterisk
>> 1.6 current with centos ( os that asterisk is developed on from my 
>> understanding )
>> with all default settings ( excluding logger.conf, dialplan and 
>> outboundproxy )
>> and I am having problems with the DTMF.
>> Both servers were installed with CentOS 5.5 and were updated last night, 
>> after
>> which I reinstalled asterisk. This did not resolve the issue.
>> I am at wit's end and do not know where to go from here. I would really 
>> appreciate
>> it if someone could give me some pointers on where to go next, what 
>> additionnal
>> debugging steps I should perform. I would also really appreciate if someone 
>> could
>> propose a solution.
>> Please help!
>> David
>> Never give up, never surrender
> 
> -- 
> Edwin Lam 
> Systems Engineer, OfficeWyze, Inc.
> Ph: +1 415 439 4988 Fax: +1 415 283 3370
> http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
> 
> 
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--
_
-- Bandwi

Re: [asterisk-users] DTMF not being sent ( RFC2833 )

[Edwin Lam]

i think i have similar problem after upgraded from 1.4.x to 1.6.2.17.
(originally upgraded to 1.8.3.2 unfortunately there were other more
pressing problems that forced me to downgraded it to 1.6.2.17)
i have a wanpipe device with 2 channels uses PRI signalling to PSTN &
the other 2 uses FXO signalling (connect to Rhino FXS channel bank).
the PRI part works fine but the FXO channels are having DTMF digits
skipped. i'm still trying to find out what's wrong with it.

On 4/23/11 8:48 AM, David wrote:

Hello,
I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple
problems with DTMF.
I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does 
IVR
and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers
communicate via SIP with RFC2833.
I setup logger.conf on both machines to display DTMF to the console. Both are
built from source.
Asterisk : spandsp, dahdi, asterisk.
Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe
I eliminated AGI, hard phones, network et al by setting up this extension :
exten => 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983
#))
in default.
The only other non default setting is in sip.conf I added a outboundproxy ( 
which
does NOT do RTP, only SIP ).
I called asterisk from my hard phone ( gxp2000 ) by dialing 22.
I see the console DTMF messages indicating the DTMF was sent or received. ( I
forgot to keep this output ).
I than watch the console DTMF output on asterisk-pri and it showed about half 
the
DTMFs. The pager that was called showed the DTMFs that appeared on the
asterisk-pri console.
So somewhere between the two machines, the DTMFs have disappeared. So I ran
TCPDump on asterisk and saw that close to half of the DTMF events were never 
sent.
tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap
I imported the file into wireshark on my local machine and confirmed that the 
dump
almost matches what I saw on asterisk-pri.
So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri.
I compared the packet scan to what I saw on asterisk-pri and noticed that 
between
1 and 3 dtmfs were missing.
Problem 2 : Asterisk-pri loses some received DTMFs.
I also noticed that some of the DTMFs coming out of asterisk had the wrong Event
Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58
seconds ) but I only pressed the button for like 1/3 of a second.
What I do not understand is that I in my final test last night was using 
asterisk
1.6 current with centos ( os that asterisk is developed on from my 
understanding )
with all default settings ( excluding logger.conf, dialplan and outboundproxy )
and I am having problems with the DTMF.
Both servers were installed with CentOS 5.5 and were updated last night, after
which I reinstalled asterisk. This did not resolve the issue.
I am at wit's end and do not know where to go from here. I would really 
appreciate
it if someone could give me some pointers on where to go next, what additionnal
debugging steps I should perform. I would also really appreciate if someone 
could
propose a solution.
Please help!
David
Never give up, never surrender


--
Edwin Lam 
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20


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Re: [asterisk-users] DTMF not being sent ( RFC2833 )

[David]

Hello,

I traced the SIP packets and saw that the only difference was that the 
DAHDI channel returns 183 Session progress ( besides the obvious 
differences such as the To and from tags in sip , session id and rtp 
ports in the SDP ).


I updated my dialplan on asterisk-pri as follows :

exten => 6,1,Progress
exten => 6,n,Wait(5)
exten => 6,n,Answer
exten => 6,n,Wait(30)

This makes the local channel behave the same as the DAHDI channel. With 
this in place, the SIP packets for both test calls are identical ( 
excluding the To header, To Tag, From Tag, SDP ports, SDP session Id and 
SDP version.


Everything else is identical. So the problem appears to be caused in the 
RTP and not in the SIP. So something about the RTP packets coming from 
the DAHDI channel on asterisk-pri makes asterisk server send invalid DTMF.


David

On 11-04-24 11:42 AM, David wrote:

I did more testing.

Here is a portion of extensions.conf on asterisk-pri:

exten => 5,1,Dial(DAHDI/g1/14186939930,30)
exten => 6,1,Answer
exten => 6,2,Wait(30)
exten => 7,1,Dial(DAHDI/g1/14186939930,30,D(132412983#))

Here is an expert from asterisk :

exten => 22,1,Dial(SIP/6@pri,30,D(132412983#))
exten => 24,1,Dial(SIP/5@pri,30,D(132412983#))

If I type "console dial 24", the DTMFs work poorly, and I see messages 
like :


[Apr 24 11:26:20] DTMF[2691]: channel.c:2907 __ast_read: DTMF end 
emulation of '1' queued on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' 
received on SIP/omnity-0004, duration 60120 ms
[Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end 
accepted with begin '1' on SIP/omnity-0004
[Apr 24 11:26:20] DTMF[2691]: channel.c:2858 __ast_read: DTMF end 
passthrough '1' on SIP/omnity-0004
[Apr 24 11:26:20] DTMF[2691]: channel.c:2874 __ast_read: DTMF begin 
'1' received on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2884 __ast_read: DTMF begin 
passthrough '1' on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' 
received on DAHDI/1-1, duration 39 ms
[Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end 
accepted with begin '1' on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2851 __ast_read: DTMF end '1' 
has duration 39 but want minimum 80, emulating on DAHDI/1-1


If I type console dial 22 on asterisk, all the DTMFs are 60ms in 
length and I get no unusually long DTMFs.


If I type console dial 7 on asterisk-pri, all the DTMFs are properly 
sent, and the remote party sees my DTMFs perfectly.


So it would seem that the bug occurs when one asterisk calls the 
second asterisk which bridges to a DAHDI channel.


My next step is too compare the SIP signalling between the two calls. 
Maybe something is different.


What I find really weird is that the DTMF is incorrectly sent from the 
first asterisk only when the second asterisk bridges to DAHDI.


Any ideas?

David

On 11-04-23 11:48 AM, David wrote:

Hello,
I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had 
multiple problems with DTMF.
I have two machines, we'll call them asterisk and asterisk-pri. 
Asterisk does IVR and asterisk-pri has a PRI card in it and connects 
to the PSTN. The two servers communicate via SIP with RFC2833.
I setup logger.conf on both machines to display DTMF to the console. 
Both are built from source.

Asterisk : spandsp, dahdi, asterisk.
Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe
I eliminated AGI, hard phones, network et al by setting up this 
extension :
exten => 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 
#))

in default.
The only other non default setting is in sip.conf I added a 
outboundproxy ( which does NOT do RTP, only SIP ).

I called asterisk from my hard phone ( gxp2000 ) by dialing 22.
I see the console DTMF messages indicating the DTMF was sent or 
received. ( I forgot to keep this output ).
I than watch the console DTMF output on asterisk-pri and it showed 
about half the DTMFs. The pager that was called showed the DTMFs that 
appeared on the asterisk-pri console.
So somewhere between the two machines, the DTMFs have disappeared. So 
I ran TCPDump on asterisk and saw that close to half of the DTMF 
events were never sent.

tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap
I imported the file into wireshark on my local machine and confirmed 
that the dump almost matches what I saw on asterisk-pri.

So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri.
I compared the packet scan to what I saw on asterisk-pri and noticed 
that between 1 and 3 dtmfs were missing.

Problem 2 : Asterisk-pri loses some received DTMFs.
I also noticed that some of the DTMFs coming out of asterisk had the 
wrong Event Duration. I had one DTMF with a duration of about 58000 ( 
I believe that's 58 seconds ) but I only pressed the button for like 
1/3 of a second.
What I do not understand is that I in my final test last night was 
us

Re: [asterisk-users] DTMF not being sent ( RFC2833 )

[David]

I did more testing.

Here is a portion of extensions.conf on asterisk-pri:

exten => 5,1,Dial(DAHDI/g1/14186939930,30)
exten => 6,1,Answer
exten => 6,2,Wait(30)
exten => 7,1,Dial(DAHDI/g1/14186939930,30,D(132412983#))

Here is an expert from asterisk :

exten => 22,1,Dial(SIP/6@pri,30,D(132412983#))
exten => 24,1,Dial(SIP/5@pri,30,D(132412983#))

If I type "console dial 24", the DTMFs work poorly, and I see messages 
like :


[Apr 24 11:26:20] DTMF[2691]: channel.c:2907 __ast_read: DTMF end 
emulation of '1' queued on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' 
received on SIP/omnity-0004, duration 60120 ms
[Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end 
accepted with begin '1' on SIP/omnity-0004
[Apr 24 11:26:20] DTMF[2691]: channel.c:2858 __ast_read: DTMF end 
passthrough '1' on SIP/omnity-0004
[Apr 24 11:26:20] DTMF[2691]: channel.c:2874 __ast_read: DTMF begin '1' 
received on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2884 __ast_read: DTMF begin 
passthrough '1' on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2802 __ast_read: DTMF end '1' 
received on DAHDI/1-1, duration 39 ms
[Apr 24 11:26:20] DTMF[2691]: channel.c:2842 __ast_read: DTMF end 
accepted with begin '1' on DAHDI/1-1
[Apr 24 11:26:20] DTMF[2691]: channel.c:2851 __ast_read: DTMF end '1' 
has duration 39 but want minimum 80, emulating on DAHDI/1-1


If I type console dial 22 on asterisk, all the DTMFs are 60ms in length 
and I get no unusually long DTMFs.


If I type console dial 7 on asterisk-pri, all the DTMFs are properly 
sent, and the remote party sees my DTMFs perfectly.


So it would seem that the bug occurs when one asterisk calls the second 
asterisk which bridges to a DAHDI channel.


My next step is too compare the SIP signalling between the two calls. 
Maybe something is different.


What I find really weird is that the DTMF is incorrectly sent from the 
first asterisk only when the second asterisk bridges to DAHDI.


Any ideas?

David

On 11-04-23 11:48 AM, David wrote:

Hello,
I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had 
multiple problems with DTMF.
I have two machines, we'll call them asterisk and asterisk-pri. 
Asterisk does IVR and asterisk-pri has a PRI card in it and connects 
to the PSTN. The two servers communicate via SIP with RFC2833.
I setup logger.conf on both machines to display DTMF to the console. 
Both are built from source.

Asterisk : spandsp, dahdi, asterisk.
Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe
I eliminated AGI, hard phones, network et al by setting up this 
extension :
exten => 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 
#))

in default.
The only other non default setting is in sip.conf I added a 
outboundproxy ( which does NOT do RTP, only SIP ).

I called asterisk from my hard phone ( gxp2000 ) by dialing 22.
I see the console DTMF messages indicating the DTMF was sent or 
received. ( I forgot to keep this output ).
I than watch the console DTMF output on asterisk-pri and it showed 
about half the DTMFs. The pager that was called showed the DTMFs that 
appeared on the asterisk-pri console.
So somewhere between the two machines, the DTMFs have disappeared. So 
I ran TCPDump on asterisk and saw that close to half of the DTMF 
events were never sent.

tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap
I imported the file into wireshark on my local machine and confirmed 
that the dump almost matches what I saw on asterisk-pri.

So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri.
I compared the packet scan to what I saw on asterisk-pri and noticed 
that between 1 and 3 dtmfs were missing.

Problem 2 : Asterisk-pri loses some received DTMFs.
I also noticed that some of the DTMFs coming out of asterisk had the 
wrong Event Duration. I had one DTMF with a duration of about 58000 ( 
I believe that's 58 seconds ) but I only pressed the button for like 
1/3 of a second.
What I do not understand is that I in my final test last night was 
using asterisk 1.6 current with centos ( os that asterisk is developed 
on from my understanding ) with all default settings ( excluding 
logger.conf, dialplan and outboundproxy ) and I am having problems 
with the DTMF.
Both servers were installed with CentOS 5.5 and were updated last 
night, after which I reinstalled asterisk. This did not resolve the issue.
I am at wit's end and do not know where to go from here. I would 
really appreciate it if someone could give me some pointers on where 
to go next, what additionnal debugging steps I should perform. I would 
also really appreciate if someone could propose a solution.

Please help!
David
Never give up, never surrender


--
_
-- Bandwidth and Colocation Provided byhttp://www.api-digital.com  --
New to Asterisk? Join us for a live introductor