Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
issue for me at this time but sill interesting. My issue though, is with
talk off on a zap channel. It seems to be getting worse or maybe my patience
is thinning. All my calls go out and come in pstn through an FXO as I do not
have high speed available here at home. My Current setup is:
Phone-->PAP2--> * --->PSTN--->Voip number to * at another
location(that has
high speed)--->send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I also
remeber seing something about relaxdtmf being set to something other than yes
or no, so I looked in chan_zap.c and found relaxdtmf in many places but it
looked to my inexperienced eye that it could only be set to 'yes' or
'no',
but i did find some mention of tonlength (at line 10858)
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen > 4000 ) || (tdp.dtmf_tonelen < 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values ignore it.
Any ideas what might happen if i increased the minimum time value? if this is
indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel => 1
<same for channel 2>
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten => s,1, NoOp(${CALLERID} time ${DATETIME});
exten => s,2, Dial(sip/677&sip/666,30,tT);
exten => <a bunch of stuff to do with agi look ups and voicemail
leave/retrieve>
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
Check.... http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.html On Sun, 18 Jun 2006, John Millican wrote:> Hello all, > I have seen some chatter again about DTMF. I see most of the talk about DTMF > around not being able to get an external IVR to recognize digits, not a big > issue for me at this time but sill interesting. My issue though, is with > talk off on a zap channel. It seems to be getting worse or maybe my patience > is thinning. All my calls go out and come in pstn through an FXO as I do not > have high speed available here at home. My Current setup is: > > Phone-->PAP2--> * --->PSTN--->Voip number to * at another location(that has > high speed)--->send to VoIP provider > > I read a post about talked about the length of the DTMFish sound. I also > remeber seing something about relaxdtmf being set to something other than yes > or no, so I looked in chan_zap.c and found relaxdtmf in many places but it > looked to my inexperienced eye that it could only be set to 'yes' or 'no', > but i did find some mention of tonlength (at line 10858) > followed that to zaptel.c (line 3357) where it said : > if ((tdp.dtmf_tonelen > 4000 ) || (tdp.dtmf_tonelen < 10 )) > return -EINVAL > Which I am guessing means unless the dtmf is between these 2 values ignore it. > Any ideas what might happen if i increased the minimum time value? if this is > indeed what this is referring to? > > > Zapata.conf: > [channels] > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > busydetect=yes > busycount=6 > echocancel=128 > echocancelwhenbridged=yes > echotraining=yes > rxgain=0 > txgain=0 > immediate=no > context=default > signalling=fxs_ks > channel => 1 > <same for channel 2> > > zaptel.conf: > loadzone = us > fxsks=1 > fxsks=2 > > extensions.conf: > exten => s,1, NoOp(${CALLERID} time ${DATETIME}); > exten => s,2, Dial(sip/677&sip/666,30,tT); > exten => <a bunch of stuff to do with agi look ups and voicemail > leave/retrieve> > > All very basic and works like a charm except for the talk off. > Thanks in advance to all, > John M > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > 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 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
John, Well I am certainly not an expert on this. I am using an SPA-3000 and I have not experienced this. I did have to go to inband on the fxo channel as rfc8322 did not work for ivr's when using Asterisk. I think you said you were using a linksys or sipura product for you fxo?? If that is the case using inband and the ulaw/alaw encoder for the fxo channel might help. Worth a try I guess. There are some rfc8322 issues that apparently will be addressed with a rewrite in the next makor version release. Doug On Mon, 19 Jun 2006, John Millican wrote:> > Doug, I read that post and unfortunately it was not a solution. I do not > believe it has to do with interstate as it happens intra state also. Is > there any way to make DTMF detection stricter, ie require a longer minimum > tone length. Assuming ( yes a dangerous practice) that the human voice will > not hold a DTMF sequence stable for very long, if I lengthen the minimum > required length I may be able to minimize the talk off. What do you think? > Any suggestions? > John M > > Doug Crompton wrote: > > Check.... > > > > > > http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.html > > > > On Sun, 18 Jun 2006, John Millican wrote: > > > Hello all, > > > I have seen some chatter again about DTMF. I see most of the talk about > > > DTMF around not being able to get an external IVR to recognize digits, > > > not a big issue for me at this time but sill interesting. My issue > > > though, is with talk off on a zap channel. It seems to be getting worse > > > or maybe my patience is thinning. All my calls go out and come in pstn > > > through an FXO as I do not have high speed available here at home. My > > > Current setup is: > > > > > > Phone-->PAP2--> * --->PSTN--->Voip number to * at another location(that > > > has high speed)--->send to VoIP provider > > > > > > I read a post about talked about the length of the DTMFish sound. I also > > > remeber seing something about relaxdtmf being set to something other than > > > yes or no, so I looked in chan_zap.c and found relaxdtmf in many places > > > but it looked to my inexperienced eye that it could only be set to 'yes' > > > or 'no', but i did find some mention of tonlength (at line 10858) > > > followed that to zaptel.c (line 3357) where it said : > > > if ((tdp.dtmf_tonelen > 4000 ) || (tdp.dtmf_tonelen < 10 )) > > > return -EINVAL > > > Which I am guessing means unless the dtmf is between these 2 values > > > ignore it. Any ideas what might happen if i increased the minimum time > > > value? if this is indeed what this is referring to? > > > > > > > > > Zapata.conf: > > > [channels] > > > callwaiting=yes > > > callwaitingcallerid=yes > > > threewaycalling=yes > > > transfer=yes > > > cancallforward=yes > > > busydetect=yes > > > busycount=6 > > > echocancel=128 > > > echocancelwhenbridged=yes > > > echotraining=yes > > > rxgain=0 > > > txgain=0 > > > immediate=no > > > context=default > > > signalling=fxs_ks > > > channel => 1 > > > <same for channel 2> > > > > > > zaptel.conf: > > > loadzone = us > > > fxsks=1 > > > fxsks=2 > > > > > > extensions.conf: > > > exten => s,1, NoOp(${CALLERID} time ${DATETIME}); > > > exten => s,2, Dial(sip/677&sip/666,30,tT); > > > exten => <a bunch of stuff to do with agi look ups and voicemail > > > leave/retrieve> > > > > > > All very basic and works like a charm except for the talk off. > > > Thanks in advance to all, > > > John M >"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 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
Doug, thanks for the help. I am using uLAW and inband every where. I have tried using 2833 and it did not appear to make any difference, better or worse. this is why I was thinking that if I could increase the minimum required time for a tone that it night help, I am just not sure where the best place top do this is. i thought I had seen a post about setting relaxdtmf to a value to actually make dtmf detection stricter but i can not seam to find anything other than 'yes' or 'no'. John Doug Crompton wrote:> John, > > Well I am certainly not an expert on this. I am using an SPA-3000 and I > have not experienced this. I did have to go to inband on the fxo channel > as rfc8322 did not work for ivr's when using Asterisk. I think you said > you were using a linksys or sipura product for you fxo?? If that is the > case using inband and the ulaw/alaw encoder for the fxo channel might > help. Worth a try I guess. There are some rfc8322 issues that apparently > will be addressed with a rewrite in the next makor version release. > > Doug > > On Mon, 19 Jun 2006, John Millican wrote: > > Doug, I read that post and unfortunately it was not a solution. I do not > > believe it has to do with interstate as it happens intra state also. Is > > there any way to make DTMF detection stricter, ie require a longer > > minimum tone length. Assuming ( yes a dangerous practice) that the human > > voice will not hold a DTMF sequence stable for very long, if I lengthen > > the minimum required length I may be able to minimize the talk off. What > > do you think? Any suggestions? > > John M > > > > Doug Crompton wrote: > > > Check.... > > > > > > > > > http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.ht > > >ml > > > > > > On Sun, 18 Jun 2006, John Millican wrote: > > > > Hello all, > > > > I have seen some chatter again about DTMF. I see most of the talk > > > > about DTMF around not being able to get an external IVR to recognize > > > > digits, not a big issue for me at this time but sill interesting. My > > > > issue though, is with talk off on a zap channel. It seems to be > > > > getting worse or maybe my patience is thinning. All my calls go out > > > > and come in pstn through an FXO as I do not have high speed available > > > > here at home. My Current setup is: > > > > > > > > Phone-->PAP2--> * --->PSTN--->Voip number to * at another > > > > location(that has high speed)--->send to VoIP provider > > > > > > > > I read a post about talked about the length of the DTMFish sound. I > > > > also remeber seing something about relaxdtmf being set to something > > > > other than yes or no, so I looked in chan_zap.c and found relaxdtmf > > > > in many places but it looked to my inexperienced eye that it could > > > > only be set to 'yes' or 'no', but i did find some mention of > > > > tonlength (at line 10858) followed that to zaptel.c (line 3357) where > > > > it said : > > > > if ((tdp.dtmf_tonelen > 4000 ) || (tdp.dtmf_tonelen < 10 )) > > > > return -EINVAL > > > > Which I am guessing means unless the dtmf is between these 2 values > > > > ignore it. Any ideas what might happen if i increased the minimum > > > > time value? if this is indeed what this is referring to? > > > > > > > > > > > > Zapata.conf: > > > > [channels] > > > > callwaiting=yes > > > > callwaitingcallerid=yes > > > > threewaycalling=yes > > > > transfer=yes > > > > cancallforward=yes > > > > busydetect=yes > > > > busycount=6 > > > > echocancel=128 > > > > echocancelwhenbridged=yes > > > > echotraining=yes > > > > rxgain=0 > > > > txgain=0 > > > > immediate=no > > > > context=default > > > > signalling=fxs_ks > > > > channel => 1 > > > > <same for channel 2> > > > > > > > > zaptel.conf: > > > > loadzone = us > > > > fxsks=1 > > > > fxsks=2> > > > > > > > extensions.conf: > > > > exten => s,1, NoOp(${CALLERID} time ${DATETIME}); > > > > exten => s,2, Dial(sip/677&sip/666,30,tT); > > > > exten => <a bunch of stuff to do with agi look ups and voicemail > > > > leave/retrieve> > > > > > > > > All very basic and works like a charm except for the talk off. > > > > Thanks in advance to all, > > > > John M > > "Those that sacrifice essential liberty to obtain a little temporary safety > deserve neither liberty nor safety." -- Ben Franklin (1759) > > **************************** > * Doug Crompton * > * Richboro, PA 18954 * > * 215-431-6307 * > * * > * doug@crompton.com * > * http://www.crompton.com * > ****************************
John, You said you were using a PAP2.. what is the FXO interface at the (far) asterisk end? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
Doug,
The interface that i dial to is at my Service provider and am not sure what
they are using. I dial out of my * box to a service provider number which is
answerd by an asterisk box that I have at another location on a high speed
cable connection, that box then dials the numberI ultimately want to reach.
I use an extensions.conf line at my home * such as:
Dial(zap/1/<my_sip_number>wwwwww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:> John,
>
> You said you were using a PAP2.. what is the FXO interface at the (far)
> asterisk end?
>
> Doug
>
> ****************************
> * Doug Crompton *
> * Richboro, PA 18954 *
> * 215-431-6307 *
> * *
> * doug@crompton.com *
> * http://www.crompton.com *
> ****************************
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
With recent versions of *, you can increase the detection time in
zapata.conf with the toneduration variable.
The default setting is:
toneduration=100
We had the same problem and solved it by increasing this to 200.
You can also increase the threshold volume for detection of DTMF by
setting VPM_DEFAULT_DTMFTHRESHOLD in the relevant zaptel wctXXXXX.c and
recompiling (though if you increase this too much you risk losing your
ability to detect DTMF at all).
Hope this helps,
Matt.
Matt, Thank you very much! I am currently running 1.2.7.1 but will be upgrading to 1.2.9.1 this week. I will try toneduration=200 first and let you/list know how well it worked. I read in zapata.conf.sample where it says: How long generated tones (DTMF and MF) will be played on the channel (in milliseconds) and did not realize that would have an effect on recognition. Thanks again, John M On Monday June 19 2006 2:58 pm, Matt King wrote:> With recent versions of *, you can increase the detection time in > zapata.conf with the toneduration variable. > > The default setting is: > > toneduration=100 > > We had the same problem and solved it by increasing this to 200. > > You can also increase the threshold volume for detection of DTMF by > setting VPM_DEFAULT_DTMFTHRESHOLD in the relevant zaptel wctXXXXX.c and > recompiling (though if you increase this too much you risk losing your > ability to detect DTMF at all). > > Hope this helps, > > Matt.
Is the PAP2 an ethernet connected device to * ? I was wondering why you were using zap if it were not an internal card? Doug On Mon, 19 Jun 2006, John Millican wrote:> Doug, > The interface that i dial to is at my Service provider and am not sure what > they are using. I dial out of my * box to a service provider number which is > answerd by an asterisk box that I have at another location on a high speed > cable connection, that box then dials the numberI ultimately want to reach. > I use an extensions.conf line at my home * such as: > Dial(zap/1/<my_sip_number>wwwwww${EXTEN}); > this works great and saves me a ton on call costs. > John > > On Monday June 19 2006 12:19 pm, Doug Crompton wrote: > > John, > > > > You said you were using a PAP2.. what is the FXO interface at the (far) > > asterisk end? > > > > Doug > > > > **************************** > > * Doug Crompton * > > * Richboro, PA 18954 * > > * 215-431-6307 * > > * * > > * doug@crompton.com * > > * http://www.crompton.com * > > **************************** > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > 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 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
Doug, The PAP2 is single ethernet and 2 fxs. I actually have a couple of these a SPA-3000, and a SPA 2102 (testing purposes). these connect to asterisk over the network and * is the house PBX set to dial a local Voip Number via the zap/PSTN, which is routed to another * box on a cable connection that dials out over the cable back to the VoIP Provider and then routed to the world. this way i can still have the lower cost of VoIP while living in my back woods New Hampshire location where High speed is not available. I currently have three PSTN lines here at the house, 1 line for dial-up and 2 plain vanilla local only lines. John On Monday June 19 2006 5:28 pm, Doug Crompton wrote:> Is the PAP2 an ethernet connected device to * ? I was wondering why you > were using zap if it were not an internal card? > > Doug > > On Mon, 19 Jun 2006, John Millican wrote: > > Doug, > > The interface that i dial to is at my Service provider and am not sure > > what they are using. I dial out of my * box to a service provider number > > which is answerd by an asterisk box that I have at another location on a > > high speed cable connection, that box then dials the numberI ultimately > > want to reach. I use an extensions.conf line at my home * such as: > > Dial(zap/1/<my_sip_number>wwwwww${EXTEN}); > > this works great and saves me a ton on call costs. > > John > > > > On Monday June 19 2006 12:19 pm, Doug Crompton wrote: > > > John, > > > > > > You said you were using a PAP2.. what is the FXO interface at the > > > (far) asterisk end? > > > > > > Doug > > > > > > **************************** > > > * Doug Crompton * > > > * Richboro, PA 18954 * > > > * 215-431-6307 * > > > * * > > > * doug@crompton.com * > > > * http://www.crompton.com * > > > **************************** > > > > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > 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 * > * * > * doug@crompton.com * > * http://www.crompton.com * > **************************** > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
John, Ok... I guess I am still not getting this. Where does the sap come in? I am running an SPA-3000 and routing all call thru *. I do not have any zap drivers installed since I do not have any internal cards. Doug On Mon, 19 Jun 2006, John Millican wrote:> Doug, > The PAP2 is single ethernet and 2 fxs. I actually have a couple of these a > SPA-3000, and a SPA 2102 (testing purposes). these connect to asterisk over > the network and * is the house PBX set to dial a local Voip Number via the > zap/PSTN, which is routed to another * box on a cable connection that dials > out over the cable back to the VoIP Provider and then routed to the world. > this way i can still have the lower cost of VoIP while living in my back > woods New Hampshire location where High speed is not available. I currently > have three PSTN lines here at the house, 1 line for dial-up and 2 plain > vanilla local only lines. > John > On Monday June 19 2006 5:28 pm, Doug Crompton wrote: > > Is the PAP2 an ethernet connected device to * ? I was wondering why you > > were using zap if it were not an internal card? > > > > Doug > > > > On Mon, 19 Jun 2006, John Millican wrote: > > > Doug, > > > The interface that i dial to is at my Service provider and am not sure > > > what they are using. I dial out of my * box to a service provider number > > > which is answerd by an asterisk box that I have at another location on a > > > high speed cable connection, that box then dials the numberI ultimately > > > want to reach. I use an extensions.conf line at my home * such as: > > > Dial(zap/1/<my_sip_number>wwwwww${EXTEN}); > > > this works great and saves me a ton on call costs. > > > John > > > > > > On Monday June 19 2006 12:19 pm, Doug Crompton wrote: > > > > John, > > > > > > > > You said you were using a PAP2.. what is the FXO interface at the > > > > (far) asterisk end? > > > > > > > > Doug > > > > > > > > **************************** > > > > * Doug Crompton * > > > > * Richboro, PA 18954 * > > > > * 215-431-6307 * > > > > * * > > > > * doug@crompton.com * > > > > * http://www.crompton.com * > > > > **************************** > > > > > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > 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 * > > * * > > * doug@crompton.com * > > * http://www.crompton.com * > > **************************** > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > 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 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
this does not make any sense. How do you dial to a service provider from your * box? Does it use PPP and IP? And then you connect to another * box that is on a cable connection that receives the call over IP and then dials out to a voip provider? How do any fxo devices come into this picture? How does a zap channel come into this picture? John Millican wrote:> Doug, > The interface that i dial to is at my Service provider and am not sure what > they are using. I dial out of my * box to a service provider number which is > answerd by an asterisk box that I have at another location on a high speed > cable connection, that box then dials the numberI ultimately want to reach. > I use an extensions.conf line at my home * such as: > Dial(zap/1/<my_sip_number>wwwwww${EXTEN}); > this works great and saves me a ton on call costs. > John > > On Monday June 19 2006 12:19 pm, Doug Crompton wrote: > >> John, >> >> You said you were using a PAP2.. what is the FXO interface at the (far) >> asterisk end? >> >> Doug >> >> **************************** >> * Doug Crompton * >> * Richboro, PA 18954 * >> * 215-431-6307 * >> * * >> * doug@crompton.com * >> * http://www.crompton.com * >> **************************** >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >