[Asterisk-Users] Cisco 7960 DTMF Problems
David Liu
dtliu at scu.edu
Thu Jan 29 23:03:48 MST 2004
Hi Eric,
I understand that both * and the SIP end-point has to have the dtmf mode the
same setting in order to work. So it has always been dtmfmode=rfc2833 in
sip.conf. My 7960 have the following DTMF setup:
dtmf_inband: 1
dtmf_outofband: avt
dtmf_db_level: 3
dtmf_avt_payload: 101
What the problem is now is. OK I get a dial tone after dialing 9 (real dial
tone from PSTN). Then I dial the phone number. (I listened carefully and
realize that say if I dialed 21120852 it would end up dialing 2111208852 or
some variations like that. Basically, when you press a button very
carefully, it may still send out 2 or 3 tones of the keys you just pressed.
If I set to INFO, nothing happens, as in whatever you press on the keypad,
it is not regenerated at the PSTN side
I tried using a Zultys ZIP2x2 and it has no DTMF problems both set at inband
or rfc2833. So it seems like a 7960 problem?
By the way, we are runing G711alaw and ver 6.1 7960 firmware.
Thanks!
David
----- Original Message -----
From: "Eric Wieling" <eric at fnords.org>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, January 29, 2004 5:56 PM
Subject: Re: [Asterisk-Users] How to delay dialing
> Set the DTMF mode to be the same on the SIP device and on Asterisk. In
> Asterisk it's dtmfmode=rfc2833 in the [sipdevice] section of sip.conf.
> BOTH Asterisk and the SIP device need to be set to the same DTMF mode.
> If the phone is sending inband DTMF and Asterisk expects rfc2833 DTMF
> it's not going to work right. If you are using inband DTMF and a
> compressed codec (basically any codec other than G711 ulaw or G711 alaw)
> then DTMF will not work right. The codec will distort continuous tones.
> You MAY have to set the DTMF mode to "info" on Asterisk and the SIP
> device. Reports on this vary.
>
> On Thu, 2004-01-29 at 19:03, David Liu wrote:
> > Thanks Eric for the suggestion.
> >
> > I currently do use that method to just give user the dialtone. However,
the
> > problem is that our 7960s somehow don't send out correct DTMF.
> >
> > For example, say if I dial 555-1212 on the Cisco 7960 keypad. I would
end
> > up connecting to 555-1221 or some other random numbers. I couldn't set
DTMF
> > to inband as that will crash Asterisk (because Voicetronix detects DTMF
on
> > the line and crashing it)
> >
> > Is there anyway to go about this?
> >
> > David
> >
> > ----- Original Message -----
> > From: "Eric Wieling" <eric at fnords.org>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Thursday, January 29, 2004 4:54 PM
> > Subject: Re: [Asterisk-Users] How to delay dialing
> >
> >
> > >
> > > I would suggest that you use exten =>
_9.,1,Dial(vpb/1-1/ww${EXTEN:1})
> > >
> > > But as I understand it the VPB channel driver does not support the "w"
> > > (wait) option on the Dial string. An alternative would be this:
> > >
> > > exten => 9,1,Dial(vpb/1-1/)
> > >
> > > Which would just connect you to the PSTN dialtone when you dialed "9".
> > >
> > > On Thu, 2004-01-29 at 18:10, David Liu wrote:
> > > > Hi there,
> > > >
> > > > I am trying to delay sending out DTMF from Voicetronix OpenLine4 to
> > > > the CO
> > > > line. The reason being is that Voicetronix sends out the DTMF too
> > > > fast even
> > > > before the line is fully established with the carrier. Usually when
> > > > dialing
> > > > an 8 digit number, only 7 digits are actually successfully heard by
> > > > the
> > > > carrier.
> > > >
> > > > Currently, my dial plan is:
> > > > exten => _9.,1,Dial(vpb/1-1/${EXTEN:1})
> > > >
> > > > Daniel said to insert a , before the numbers. I am not too sure
where
> > > > to
> > > > insert it. I tried
> > > > exten => _9.,1,Dial(vpb/1-1/,${EXTEN:1}) and that seems to be cause
a
> > > > parsing error.
> > > >
> > > > Anybody has any ideas for a hack?
> > > >
> > > > David
> > > >
> > > >
> > > >
______________________________________________________________________
> > > > This message has been scaned for viruses. This means that
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> > > >
> > > > Sanitizer (start="1075422274"):
> > > > Part (pos="2797"):
> > > > SanitizeFile (filename="unnamed.txt", mimetype="text/plain"):
> > > > Match (names="unnamed.txt", rule="1"):
> > > > Enforced policy: accept
> > > >
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> > > > Match (names="unnamed.html", rule="2"):
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> > > > Scan succeeded, file is clean.
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> > > > Note: Styles and layers give attackers many tools to fool the
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> > > > -
> >
http://archives.indenial.com/hypermail/bugtraq/2001/January2001/0512.html
> > > > Rewrote HTML tag: >>_style_0 _/STYLE_<<
> > > > as: >>_DANGEROUS_style_0 _/STYLE_<<
> > > > Total modifications so far: 1
> > > >
> > > >
> > > >
> > > > Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.81 2003/12/17
> > > > 12:49:44 bre Exp $
> > > >
> > > --
> > > Go to http://www.digium.com/index.php?menu=documentation and look at
> > > the "Unofficial Links" section. This section has links to a wide
> > > variety of 3rd party Asterisk related pages. My page is the
> > > "Asterisk Resource Pages".
> > >
> > > BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643
> > >
> > >
> > > _______________________________________________
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> > > Asterisk-Users at lists.digium.com
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
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> --
> Go to http://www.digium.com/index.php?menu=documentation and look at
> the "Unofficial Links" section. This section has links to a wide
> variety of 3rd party Asterisk related pages. My page is the
> "Asterisk Resource Pages".
>
> BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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