[Asterisk-Users] DTMF feedthru again...

[Doug Crompton]

Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!

When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.

I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.

What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.

I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.

How can I fix this???

Doug

****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Tom Vile]

try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Doug Crompton]

Tried that makes no difference. Did it for you? What DMF method(s) are you
using. Looking at a goggle search yields lots of talk on this but no real
solution. Apparently there is an rfc2833 issue and * is working on it???
Also it appears the codec used might be an issue. This is a serious
problem in my book. It precludes me from using any DTMF over PSTN with *
at this point.

Any further help or explanation would be appreciated.

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)

****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Tom Vile]

Using AVT in my sipura with above settings and it work fine going out
the PSTN. There was an issue a while back with an older version of
Asterisk with one of my providers but it has been fine since the
upgrade. I also use ulaw for calls.

Post by Doug Crompton
Tried that makes no difference. Did it for you? What DMF method(s) are you
using. Looking at a goggle search yields lots of talk on this but no real
solution. Apparently there is an rfc2833 issue and * is working on it???
Also it appears the codec used might be an issue. This is a serious
problem in my book. It precludes me from using any DTMF over PSTN with *
at this point.
Any further help or explanation would be appreciated.

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Doug Crompton]

Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.

I can't be the only one having this problem!

Doug

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)

****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Mike Lynchfield]

in Fact we saw similar problems with all sipura products.

We think its a default value thats not quite right for the north american
market, these units are built and tested in asia mostly.

one simple test to check it out is call this number
www.nextwavetitaniumplus.com Toll-Free Account Information Line:
888-252-9535
it just seemd that even the cisco is not passing the dtmf ..

Can anyone confirm ?

Post by Doug Crompton
Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call

completion.

Post by Tom Vile

Post by Doug Crompton
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed

call,

Post by Tom Vile

Post by Doug Crompton
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags

fixed

Post by Tom Vile

Post by Doug Crompton
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Matt]

I've never had this problem with the SPA-2002s... but you could always
set DTMF to 'inband' rather then RFC.. that will cause whatever goes
out of the phone to go right across with no transcoding.

Post by Mike Lynchfield
in Fact we saw similar problems with all sipura products.
We think its a default value thats not quite right for the north american
market, these units are built and tested in asia mostly.
one simple test to check it out is call this number
888-252-9535
it just seemd that even the cisco is not passing the dtmf ..
Can anyone confirm ?

Post by Doug Crompton
Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call

completion.

Post by Doug Crompton

Post by Tom Vile

Post by Doug Crompton
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed

call,

Post by Doug Crompton

Post by Tom Vile

Post by Doug Crompton
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags

fixed

Post by Doug Crompton

Post by Tom Vile

Post by Doug Crompton
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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Asterisk-Users mailing list

http://lists.digium.com/mailman/listinfo/asterisk-users

Post by Doug Crompton

Post by Tom Vile
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list

http://lists.digium.com/mailman/listinfo/asterisk-users

Post by Doug Crompton
"Those that sacrifice essential liberty to obtain a little temporary

safety

Post by Doug Crompton
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list

http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
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[Tom Vile]

But access to voicemail in Asterisk will not work with inband.

Post by Matt
I've never had this problem with the SPA-2002s... but you could always
set DTMF to 'inband' rather then RFC.. that will cause whatever goes
out of the phone to go right across with no transcoding.

Post by Mike Lynchfield
in Fact we saw similar problems with all sipura products.
We think its a default value thats not quite right for the north american
market, these units are built and tested in asia mostly.
one simple test to check it out is call this number
888-252-9535
it just seemd that even the cisco is not passing the dtmf ..
Can anyone confirm ?

Post by Doug Crompton
Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call

completion.

Post by Doug Crompton

Post by Tom Vile

Post by Doug Crompton
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed

call,

Post by Doug Crompton

Post by Tom Vile

Post by Doug Crompton
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags

fixed

Post by Doug Crompton

Post by Tom Vile

Post by Doug Crompton
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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Asterisk-Users mailing list

http://lists.digium.com/mailman/listinfo/asterisk-users

Post by Doug Crompton

Post by Tom Vile
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list

http://lists.digium.com/mailman/listinfo/asterisk-users

Post by Doug Crompton
"Those that sacrifice essential liberty to obtain a little temporary

safety

Post by Doug Crompton
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list

http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
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_______________________________________________
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[Doug Crompton]

AVT??? I have ulaw allowed (only) - When you call your cell via
pstn/spa-3000/* and listen on both while pressing dtmf do you hear good
clean tones of enough duration to allow detection, in both directions?

Do you access DTMF required services over pstn, like banking, vm, etc
from local * system?

Doug

Post by Tom Vile
Using AVT in my sipura with above settings and it work fine going out
the PSTN. There was an issue a while back with an older version of
Asterisk with one of my providers but it has been fine since the
upgrade. I also use ulaw for calls.

Post by Doug Crompton
Tried that makes no difference. Did it for you? What DMF method(s) are you
using. Looking at a goggle search yields lots of talk on this but no real
solution. Apparently there is an rfc2833 issue and * is working on it???
Also it appears the codec used might be an issue. This is a serious
problem in my book. It precludes me from using any DTMF over PSTN with *
at this point.
Any further help or explanation would be appreciated.

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)

****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Tom Vile]

AVT is the DTMF setting. Yes, I hear the tones fine. Play with the
dtmf playback length and adjust it, also play with the dtmf playback
level its default is .16 I believe.

Post by Doug Crompton
AVT??? I have ulaw allowed (only) - When you call your cell via
pstn/spa-3000/* and listen on both while pressing dtmf do you hear good
clean tones of enough duration to allow detection, in both directions?
Do you access DTMF required services over pstn, like banking, vm, etc
from local * system?
Doug

Post by Tom Vile
Using AVT in my sipura with above settings and it work fine going out
the PSTN. There was an issue a while back with an older version of
Asterisk with one of my providers but it has been fine since the
upgrade. I also use ulaw for calls.

Post by Doug Crompton
Tried that makes no difference. Did it for you? What DMF method(s) are you
using. Looking at a goggle search yields lots of talk on this but no real
solution. Apparently there is an rfc2833 issue and * is working on it???
Also it appears the codec used might be an issue. This is a serious
problem in my book. It precludes me from using any DTMF over PSTN with *
at this point.
Any further help or explanation would be appreciated.

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call completion.
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call,
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

"Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety." -- Ben Franklin (1759)
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
_______________________________________________
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[Doug Crompton]

The only thing I have found that tends to point to an * problem is

http://bugs.digium.com/view.php?id=6667

It is a long read and I have no ideas what the disposition is. It was a
discussion back in late March. This seems to apply to all or many SIP
connected devices and around implementation of the RFC. Someone had an rtp
patch which they claimed worked and it later was taken back out. Digium is
working on it. These are a few of the things I get from the thread.

Doug

Post by Mike Lynchfield
in Fact we saw similar problems with all sipura products.
We think its a default value thats not quite right for the north american
market, these units are built and tested in asia mostly.
one simple test to check it out is call this number
888-252-9535
it just seemd that even the cisco is not passing the dtmf ..
Can anyone confirm ?

Post by Doug Crompton
Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call

completion.

Post by Tom Vile

Post by Doug Crompton
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed

call,

Post by Tom Vile

Post by Doug Crompton
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags

fixed

Post by Tom Vile

Post by Doug Crompton
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
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--
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Baldwin Technology Solutions, Inc
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www.baldwintechsolutions.com
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****************************
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* Richboro, PA 18954 *
* 215-431-6307 *
* *
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****************************
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****************************
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* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Doug Crompton]

Two other Digium bug reports on this issue. It sure looks like it is an *
issue and rather complex??? Any hope for a solution??

http://bugs.digium.com/view.php?id=5970
http://bugs.digium.com/view.php?id=6027

Post by Mike Lynchfield
in Fact we saw similar problems with all sipura products.
We think its a default value thats not quite right for the north american
market, these units are built and tested in asia mostly.
one simple test to check it out is call this number
888-252-9535
it just seemd that even the cisco is not passing the dtmf ..
Can anyone confirm ?

Post by Doug Crompton
Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug

Post by Tom Vile
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.

Post by Doug Crompton
Ok trying this again... is there anyone using the SPA-3000 with * ????
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after call

completion.

Post by Tom Vile

Post by Doug Crompton
Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed

call,

Post by Tom Vile

Post by Doug Crompton
and pressing a DTMF button on the opposing phone results in an audible
click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am
not using any dial flags.
I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags

fixed

Post by Tom Vile

Post by Doug Crompton
it but that does not now seem to be the case.
How can I fix this???
Doug
****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
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--
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Baldwin Technology Solutions, Inc
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www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
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****************************
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* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
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****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Vahan Yerkanian]

Post by Doug Crompton
Two other Digium bug reports on this issue. It sure looks like it is an *
issue and rather complex??? Any hope for a solution??
http://bugs.digium.com/view.php?id=5970
http://bugs.digium.com/view.php?id=6027

It is an * issue, tested and confirmed. Makes any PSTN side IVR
application unaccessible. It is there for the past 2 years. Current
rfc2833 implementation in Asterisk requires a total rewrite to fix this.
This issue was raised several times, with some answers being an attempt
to challenge Sipura's own rfc2833 implementation validity or a promise
to fix or submit-your-fix kind of answers. Several possible fixes were
posted but never made it to the trunk. If you can leave * out of the
path, you wont have any DTMF problems, as soon as * is in the voice
path, the dtmf packet sequencing bug manifests itself.


You have a choice of disabling all codecs except alaw/ulaw and sticking
to the inband dtmf option on both sipura and asterisk sides.

HTH,
Vahan

[Doug Crompton]

Ok what a pain the * ...... Inband works for now. It probably did not work
before because I did not specifically only allow ulaw when I did this. So
I set inband in four places - line1 and pstn in Spa-3000 and in my fxs1
and fx0 contexts in sip.config.

The other gotcha here which I still think is (also) an * problem is that
if you set any features on "tTwW" etc it filters the character and it does
not send it out over PSTN... such that if I set xfer to *7 when you hit
only the * it never goes out over PSTN even after the 500ms feature key
wait. This might be the only way it can for inband but why couldn't *
filter the first defined feature character and if the second were not
entered in feature digit timeout, send that out over PSTN.

This is all confirmed by actually calling myself and listening. I think
there are some serious issues here. Maybe more then one. It would be nice
to have them fixed.

Doug

****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Don Pobanz]

Post by Doug Crompton
The other gotcha here which I still think is (also) an * problem is that
if you set any features on "tTwW" etc it filters the character and it does
not send it out over PSTN...

Me too! :)
We have a server that we wanted the ability to transfer outgoing calls
and so included the ,T option with dial. Once we did that we could not
make remote IVR systems recognize dtmf tones. Our trunks are PRI ISDN.
We are using a T410P card connected to a channel bank for analog phones.
(No SIP in this case) Removing the outgoing transfer fixed the IVR
issues. This is definitely an asterisk issue.

Don Pobanz

[Doug Crompton]

What do you mean you cannot access VM?

I am totally inband here on SPA3000 to fix the DTMF feedthru problem and
there is no problem with * VM. I can access it from local analog phone or
I can call in and '*' it to get PW prompt.

Doug

Post by Tom Vile
But access to voicemail in Asterisk will not work with inband.
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
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"Those that sacrifice essential liberty to obtain a little temporary safety
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****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Tom Vile]

I meant info not inband. Sorry.

Post by Doug Crompton
What do you mean you cannot access VM?
I am totally inband here on SPA3000 to fix the DTMF feedthru problem and
there is no problem with * VM. I can access it from local analog phone or
I can call in and '*' it to get PW prompt.
Doug

Post by Tom Vile
But access to voicemail in Asterisk will not work with inband.
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: [phone] x205
Fax: [phone]
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"Those that sacrifice essential liberty to obtain a little temporary safety
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****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* http://www.crompton.com *
****************************
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[Doug Crompton]

Ok well I am not crazy! This seems like such an important issue I am not
sure why it has lasted for so long. DTMF is the backbone of everything we
do here. Without it we would not have calls!! At least get the DTMF stuff
right. I feel a little guilty complaining since this is free but it is
also used in some high level demanding situations (not mine) and thus is
not a trivial issue.

Doug

Post by Don Pobanz

Post by Doug Crompton
The other gotcha here which I still think is (also) an * problem is that
if you set any features on "tTwW" etc it filters the character and it does
not send it out over PSTN...

Me too! :)
We have a server that we wanted the ability to transfer outgoing calls
and so included the ,T option with dial. Once we did that we could not
make remote IVR systems recognize dtmf tones. Our trunks are PRI ISDN.
We are using a T410P card connected to a channel bank for analog phones.
(No SIP in this case) Removing the outgoing transfer fixed the IVR
issues. This is definitely an asterisk issue.
Don Pobanz
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"Those that sacrifice essential liberty to obtain a little temporary safety
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****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Eric "ManxPower" Wieling]

Post by Doug Crompton
Ok well I am not crazy! This seems like such an important issue I am not
sure why it has lasted for so long. DTMF is the backbone of everything we
do here. Without it we would not have calls!! At least get the DTMF stuff
right. I feel a little guilty complaining since this is free but it is
also used in some high level demanding situations (not mine) and thus is
not a trivial issue.

In my experience, PSTN DTMF problems are usually a volume issue. Play
with the receive and transmit gains on the SIPura FXO port.

Of course when doing SIP you need your DTMF mode (RFC2833 is what I
recommend) correct before you start trying to fix the PSTN DTMF issues.

[Nick Hoffman]

Post by Eric "ManxPower" Wieling

Post by Doug Crompton
Ok well I am not crazy! This seems like such an important issue I am
not sure why it has lasted for so long. DTMF is the backbone of
everything we do here. Without it we would not have calls!! At least
get the DTMF stuff right. I feel a little guilty complaining since
this is free but it is also used in some high level demanding
situations (not mine) and thus is not a trivial issue.

In my experience, PSTN DTMF problems are usually a volume issue. Play
with the receive and transmit gains on the SIPura FXO port.
Of course when doing SIP you need your DTMF mode (RFC2833 is what I
recommend) correct before you start trying to fix the PSTN DTMF issues.

Hi guys. I've just read through this thread and am a bit confused. Is this
DTMF issue related only to using Sipura/Linksys ATAs with Asterisk, or
does it pertain to every device that connects to Asterisk, including
FXS/FXO PCI cards?

Cheers,
-- Nick
e: ***@altcall.com
p: [phone]
f: [phone]

If you receive this email by mistake, please notify us and do not make any
use of the email. We do not waive any privilege, confidentiality or
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[Martin Joseph]

Post by Doug Crompton
Ok well I am not crazy! This seems like such an important issue I am not
sure why it has lasted for so long. DTMF is the backbone of everything we
do here. Without it we would not have calls!! At least get the DTMF stuff
right. I feel a little guilty complaining since this is free but it is
also used in some high level demanding situations (not mine) and thus is
not a trivial issue.

Actually I don't think this issue is exactly as it seems. I have also
seem the issue you are describing. But at the same time on the same
server I have other ATA's connected that are using RFC2833 and they
function properly (ie DTMF words AOK).

I think the * VM issue is actually with the SIP info method for DTMF,
not inband.

Another though I have is that if you are calling from the FXS through
the FXO on your SPA3000 (whem lot of contractions), shouldn't you be
able to reinvite and get * out of the media path entirely? If the DTMF
still fails it's not Asterisks fault.

I do also have an issue with RFC2833 not working properly on one ATA
(wellgate 3701a), but I kind of presume that to be wellgates
implementation issue.

I would think that if this DTMF issue was as plainly obvious as this
thread suggests it would have been fixed long ago? I hope?

Marty

[Doug Crompton]

Nick,

Here is what I have been able to ascertain in the past few days.

There is a known problem in * regarding RFC2833. While this shows up, in
my case with the Sipura SPA-3000, I am sure it applies to many other
devices. Many may not know they even have a problem if they do not use
DTMF codes on an established connection. An easy way to fond out is to
call your * from your cell phone and with one phone on one ear, and one on
the other, hit keys on each phone while listening on the other. You should
be able to hear at least .5 seconds or so of DTMF. In he case where it is
not working you hear a clicking sound and just a hint of DTMF or none
at all.

I solved this for now by going to inband between * and my SPA-3000 and
limiting the codec to ulaw.

Yet another * problem that is possibly related is that if you activate any
features, "TtWw" etc. in your incoming or outgoing dial plan to the pstn
using the device, in my case the SPA-3000, even if you are using inband, *
will filter the first DTMF character that is defined. Thus if you set
transfer to *9, and you have a "T" and/or "t" in your dial plan to pstn,
the '*' key will be filtered and not heard on the other end in much the
same way it sounds when using rfc2833 without features enabled. The
correct behavior, assuming you do not want the keys to be heard at the
distant end, would be to mute the key and if a second key were not
selected in the timeout period then send that keys DTMF. This would
probably be difficult to achieve with inband but if you were using a non
inband method I expect the digit could be buffered and delayed just
enough to 'pick off' and mute/take action or send the DTMF.

Sorry for the long winded reply. So much has been written on this. Goggle
is full of complaints with few if any answers. Often the device gets
blamed and as in the case of the SPA-3000 there are hundreds of settings
and at least a half dozen that might effect this, so there is always
someone who tells you to do this or that.

I just wish that Digium would have made a statement saying they are aware
of the problem and are working on it. I did kind of dig that out of the
Digium archives but it seems to have ended back in March. If the problem
were just published as known and a possible temporary fix it would save a
lot of people the grief of having to reinvent the wheel and find out all
over again what is wrong.

Doug

Post by Nick Hoffman

Post by Eric "ManxPower" Wieling

Post by Doug Crompton
Ok well I am not crazy! This seems like such an important issue I am
not sure why it has lasted for so long. DTMF is the backbone of
everything we do here. Without it we would not have calls!! At least
get the DTMF stuff right. I feel a little guilty complaining since
this is free but it is also used in some high level demanding
situations (not mine) and thus is not a trivial issue.

In my experience, PSTN DTMF problems are usually a volume issue. Play
with the receive and transmit gains on the SIPura FXO port.
Of course when doing SIP you need your DTMF mode (RFC2833 is what I
recommend) correct before you start trying to fix the PSTN DTMF issues.

Hi guys. I've just read through this thread and am a bit confused. Is this
DTMF issue related only to using Sipura/Linksys ATAs with Asterisk, or
does it pertain to every device that connects to Asterisk, including
FXS/FXO PCI cards?
Cheers,
-- Nick
p: [phone]
f: [phone]
If you receive this email by mistake, please notify us and do not make any
use of the email. We do not waive any privilege, confidentiality or
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"Those that sacrifice essential liberty to obtain a little temporary safety
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****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************

[Doug]

Post by Doug Crompton
Nick,
Here is what I have been able to ascertain in the past few days.

Hey Doug,

I went through a similar hassle with Linksys PAP2's a few
weeks back. I went round and round with Linksys tech support.
They said use "AVT". I tried it and it didn't work.

What did work both inbound and outbound was "INFO" (sometimes
called "SIP INFO"). This did the trick for us. Try it and
let us know.

[Kevin P. Fleming]

Post by Doug Crompton
I just wish that Digium would have made a statement saying they are
aware
of the problem and are working on it. I did kind of dig that out of
the
Digium archives but it seems to have ended back in March. If the

We have done exactly that. It is being worked on, and Asterisk 1.4 will have a vastly improved RFC-2833 implementation. However, for most people (90% or more), the existing implementation works just fine and they have no complaints.

The problem you mentioned regarding pressing single keys that are lead-ins to feature codes is something separate, and may very well be a legitimate bug. I've just reviewed the code in Asterisk 1.2.x and it appears to do the right then when a feature DTMF sequence 'times out' and is not completed (that is, it sends the already-pressed DTMF digits to the other party), but if you can reproduce the problem and provide a complete console trace please feel free to open a bug on bugs.digium.com.

[Gabriel Afana]

Post by Kevin P. Fleming
We have done exactly that. It is being worked on, and Asterisk 1.4 will

have a vastly improved RFC-2833 implementation. However, for most people
(90% or more), the existing implementation works just fine and they have no
complaints.

When is 1.4 expected to be released?

- Gabe

[Doug Crompton]

Ok Thank you Kevin. The situation with the lead-in key is a little hard to
get a handle on with the RFC-8233 being broken. I have to use inband and
doing the feature key stuff with inband would be a little tuff. Is that
suppose to work? If you did not care if * muted the keys it would be a
little easier. Just monitor and take action if the right keys were pressed
in the appropriate time. It is the muting that gets tricky.

Doug

Post by Kevin P. Fleming
We have done exactly that. It is being worked on, and Asterisk 1.4 will
have a vastly improved RFC-2833 implementation. However, for most people
(90% or more), the existing implementation works just fine and they have
no complaints.
The problem you mentioned regarding pressing single keys that are
lead-ins to feature codes is something separate, and may very well be a
legitimate bug. I've just reviewed the code in Asterisk 1.2.x and it
appears to do the right then when a feature DTMF sequence 'times out'
and is not completed (that is, it sends the already-pressed DTMF digits
to the other party), but if you can reproduce the problem and provide a
complete console trace please feel free to open a bug on
bugs.digium.com.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
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[Doug Crompton]

Doug,

Well I am told that INFO does not work with * VM. I never tried it....
In any event the answer here was to use INBAND for the fxo (PSTN tab) and
in sip.config for fxo also set INBAND and allow only ulaw. I was able to
leave the fxs1 channel at auto on the Line 1 tab and rfc8233 in sip.c -
also you cannot use any feature flags for the PSTN channel, either in or
out.

This is for the SPA-3000

Doug

Post by Doug

Post by Doug Crompton
Nick,
Here is what I have been able to ascertain in the past few days.

Hey Doug,
I went through a similar hassle with Linksys PAP2's a few
weeks back. I went round and round with Linksys tech support.
They said use "AVT". I tried it and it didn't work.
What did work both inbound and outbound was "INFO" (sometimes
called "SIP INFO"). This did the trick for us. Try it and
let us know.

****************************
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* ***@crompton.com *
* http://www.crompton.com *
****************************