[Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain

[Innocent Evil]

Other than  below:

Got RTP packet from x.y.z.sip_phone:10006 (type 18, seq 25407, ts 191360,
len 40)
Sent RTP packet to x.y.z.asterisk:10006 (type 18, seq 63928, ts 193744, len
20)

I dont see any message while sending digits.




> -----Original Message-----
> From: brian.west at mac.com
> Sent: Fri, 19 Aug 2005 14:33:14 -0500
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's
> VoiceMailMain
>
> If you can get an rtp debug while your pressing digits I can see if
> maybe your device is sending the digits incorrectly.
>
> /b
>
> On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote:
>
> > my sip phone have dtmf relay: rfc2833
> > asterisk sip.conf have dtmf relay: rfc2833 in associated context.
> >
> > I tried with Inband.. but g729 doesn't support it. I have g729
> > liscence from
> > digium
> > I havn't try with INFO yet.
> >
> > I prefer to have rfc2833 as dtmf relay.
> >
> > Is there any other thing that can cause this issue?
> >
> > Thanks,
> >
> >
> >
> >
> >> -----Original Message-----
> >> From: jgault at winworld.cc
> >> Sent: Fri, 19 Aug 2005 14:21:27 -0400
> >> To: asterisk-users at lists.digium.com
> >> Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's
> >> VoiceMailMain
> >>
> >> It sounds to me like an issue of transmitting DTMF tones from the SIP
> >> phones.
> >>
> >> There are several methods that can be used to accomplish DTMF from
> >> SIP
> >> phones.  Of course, you may ask why it isn't just sent as audio
> >> (like a
> >> regular POTS phone would.)  What happens if you are using a SIP
> >> phone,
> >> hold down the number 4 button for two seconds (so it sends 2 seconds
> >> worth of DTMF on the audio stream) and there is some packet loss
> >> during
> >> that time?  You'll have an audio dropout (thus, tone followed by
> >> brief
> >> silence and tone again.)  The remote end will see this as two
> >> tones, not
> >> one, which obviously can cause undesired results (and is why it's
> >> not a
> >> good idea to send DTMF in the audio stream.)
> >>
> >> That being said, look in your sip.conf for a dtmfmode parameter.  You
> >> can use inband (in the audio stream, not recommended), RFC2833, or
> >> SIP
> >> INFO.  Your SIP phone should also allow you to set how DTMF is sent
> >> (although it may not support all of these formats.)  Preferably, use
> >> RFC2833 or SIP INFO.  Find a setting that is available on your
> >> phone and
> >> on *, and make sure they're set to match.  Once you do that, it
> >> should
> >> work.
> >>
> >>           Jeremy
> >>
> >> Innocent Evil wrote:
> >>
> >>
> >>> Hi,
> >>>
> >>> I am using Asterisk cmd VoiceMailMain to manage voice mail.
> >>> Problem is, voice mail box can't read password sent from SIP
> >>> phone, but
> >>>
> >> I
> >>
> >>> don't have any problem to read password from the handset attached
> >>> to my
> >>> asterisk box.
> >>>
> >>> Your help will be greatly appreciated.
> >>>
> >>> Thanks,_______________________________________________
> >>> 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
> >>>
> >>>
> >>>
> >>
> >> --
> >> Jeremy Gault, KD4NED    <jgault at winworld.cc>
> >> Network Administrator, WinWorld Corporation
> >> Member: Bradley County ACS/RACES/SkyWarn
> >> voice: +1.423.473.8084  fax: +1.423.472.9465
> >> fwd: 461771             msn msgr: jgault at winworld.cc
> >>
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> >>
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