Low, Adam
2003-Aug-22 09:05 UTC
[Asterisk-Users] DTMF tones not long enough on out going call s
Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !?> -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF tones not long enough on out > going calls > > > DTMF tones are not long enough on out going calls, when I'm > using either > "info" or rfc2833. Does anyone know if the tone length value > is in rtp.c > or chan_sip.c ? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
Eric Wieling
2003-Aug-22 09:17 UTC
[Asterisk-Users] DTMF tones not long enough on out going call s
Have someone using a SIP device with RFC2833 signaling call you, now have the press and hold down one of the dialing keys. You'll hear a short tone then nothing. On Fri, 2003-08-22 at 11:05, Low, Adam wrote:> Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > > > -----Original Message----- > > From: James Sizemore [mailto:james@deny.org] > > Sent: 22 August 2003 17:33 > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] DTMF tones not long enough on out > > going calls > > > > > > DTMF tones are not long enough on out going calls, when I'm > > using either > > "info" or rfc2833. Does anyone know if the tone length value > > is in rtp.c > > or chan_sip.c ? > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ********* DISCLAIMER ********* > > This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)
Adam Roach
2003-Aug-22 11:08 UTC
[Asterisk-Users] DTMF tones not long enough on out going call s
I'll point out that the same applies in general to many commercial PBXes. I can verify from years of personal experience, for example, that the Ericsson MD110 (probably the most popular PBX in Europe) exhibits precisely the same behavior. /a> -----Original Message----- > From: Eric Wieling [mailto:eric@fnords.org] > Sent: Friday, August 22, 2003 11:17 > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] DTMF tones not long enough on out going > call s > > > Have someone using a SIP device with RFC2833 signaling call you, now > have the press and hold down one of the dialing keys. You'll hear a > short tone then nothing. > > On Fri, 2003-08-22 at 11:05, Low, Adam wrote: > > Maybe its just me but I find this question a little > confusing, the tone duration should have no impact on tone > recognition and typically in my experience the duration of > the tone is defined by how long the user holds down the button !? > > > > > -----Original Message----- > > > From: James Sizemore [mailto:james@deny.org] > > > Sent: 22 August 2003 17:33 > > > To: asterisk-users@lists.digium.com > > > Subject: [Asterisk-Users] DTMF tones not long enough on out > > > going calls > > > > > > > > > DTMF tones are not long enough on out going calls, when I'm > > > using either > > > "info" or rfc2833. Does anyone know if the tone length value > > > is in rtp.c > > > or chan_sip.c ? > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ********* DISCLAIMER ********* > > > > This message and any attachment are confidential and may be > privileged or otherwise protected from disclosure and may > include proprietary information. If you are not the intended > recipient, please telephone or email the sender and delete > this message and any attachment from your system. If you are > not the intended recipient you must not copy this message or > attachment or disclose the contents to any other person > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
James Sizemore
2003-Aug-22 11:16 UTC
[Asterisk-Users] DTMF tones not long enough on out going call s
Thats the way it supposed to work, but the tone sending is broken when using SIP right now. The tone is so short that a lot of IVRs don't get enough sound to detect the sound as a tone. Low, Adam wrote:>Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > > > >>-----Original Message----- >>From: James Sizemore [mailto:james@deny.org] >>Sent: 22 August 2003 17:33 >>To: asterisk-users@lists.digium.com >>Subject: [Asterisk-Users] DTMF tones not long enough on out >>going calls >> >> >>DTMF tones are not long enough on out going calls, when I'm >>using either >>"info" or rfc2833. Does anyone know if the tone length value >>is in rtp.c >>or chan_sip.c ? >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > > >********* DISCLAIMER ********* > >This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > >
Low, Adam
2003-Aug-22 12:03 UTC
[Asterisk-Users] DTMF tones not long enough on out going call s
Hmmm, interesting I was not aware of this. My experience evolves from voice switches (Nortel DMS, Lucent EXS) rather than PBX's. I am using inband DTMF within my setup as I had problems (no DTMF recognition on PSTN calls) with RFC2833 when dialing through an AS5300 and onto a DMS100. What is the reason for this? Is the PBX actually cancelling out the DTMF tone after it itself recognises the tone? Does it also effect the DTMF tones received over a B channel from the PSTN? -----Original Message----- From: Adam Roach To: 'asterisk-users@lists.digium.com' Sent: 22/08/03 20:08 Subject: RE: [Asterisk-Users] DTMF tones not long enough on out going call s I'll point out that the same applies in general to many commercial PBXes. I can verify from years of personal experience, for example, that the Ericsson MD110 (probably the most popular PBX in Europe) exhibits precisely the same behavior. /a> -----Original Message----- > From: Eric Wieling [mailto:eric@fnords.org] > Sent: Friday, August 22, 2003 11:17 > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] DTMF tones not long enough on out going > call s > > > Have someone using a SIP device with RFC2833 signaling call you, now > have the press and hold down one of the dialing keys. You'll hear a > short tone then nothing. > > On Fri, 2003-08-22 at 11:05, Low, Adam wrote: > > Maybe its just me but I find this question a little > confusing, the tone duration should have no impact on tone > recognition and typically in my experience the duration of > the tone is defined by how long the user holds down the button !? > > > > > -----Original Message----- > > > From: James Sizemore [mailto:james@deny.org] > > > Sent: 22 August 2003 17:33 > > > To: asterisk-users@lists.digium.com > > > Subject: [Asterisk-Users] DTMF tones not long enough on out > > > going calls > > > > > > > > > DTMF tones are not long enough on out going calls, when I'm > > > using either > > > "info" or rfc2833. Does anyone know if the tone length value > > > is in rtp.c > > > or chan_sip.c ? > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ********* DISCLAIMER ********* > > > > This message and any attachment are confidential and may be > privileged or otherwise protected from disclosure and may > include proprietary information. If you are not the intended > recipient, please telephone or email the sender and delete > this message and any attachment from your system. If you are > not the intended recipient you must not copy this message or > attachment or disclose the contents to any other person > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
Adam Roach
2003-Aug-24 09:21 UTC
[Asterisk-Users] DTMF tones not long enough on out going call s
Low, Adam [mailto:ALow@Prioritytelecom.com] writes: ...> What is the reason for this? Is the PBX actually cancelling > out the DTMF tone after it itself recognises the tone?No; the phone is sending the information symbolically instead of encoding it using a traditional codec. It's up to the recipient (Asterisk in this case) to interpret them correctly.> Does > it also effect the DTMF tones received over a B channel from the PSTN?Not over a B channel, since those *are* in-band. They're encoded just like voice is. On the other hand, Q.931 specifies a D-channel encoding of keypresses as well. I'm not familiar enough with actual ISDN terminal implementations to know whether they tend to send in-band, out-of-band, or both. I would guess that behavior at Asterisk would depend heavily on this factor. (Note that I don't consider myself an ISDN expert, and my knowledge in this area is only what I can recall after four years.) /a