[Asterisk-Users] DTMF issues with SIPPhone?

[Tarpo, Louie]

We encountered the same issue.  change dtmfmode=rfc2833 to dtmfmode=inband and make sure you're using a ulaw connection.  If you use a lossy codec, it will scramble the DTMF tones.


Your config would change like so,

[sipphone]
type=peer
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
fromuser=1747xxxxxxx
username=1747xxxxxxx
password=xxxxx
context=fromsipphone
dtmfmode=inband
canreinvite=no
disallow=all
allow=ulaw

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 10:27 AM
To: Asterisk-Users at lists.digium.com
Subject: [Asterisk-Users] DTMF issues with SIPPhone?


Does anyone else have DTMF issues with SIPPhone?  When calling into my
DID, and entering, say, 1002.  Sometimes it will recognize it properly
(rarely), other times it will receive something different.  Such as,
1102 or 1000, etc.  Has anyone else been having these issues?  I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:

[sipphone]
type=peer
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
fromuser=1747xxxxxxx
username=1747xxxxxxx
password=xxxxx
context=fromsipphone
dtmfmode=rfc2833
canreinvite=no


Any ideas?  Am I doing something wrong?

Thanks!
-JD-
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