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

[Jeremy Gault]

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:
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>

-- 
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