In my setup, I am using TDM Wildcards analog connections and the Asterisk PBX
box does the converting to my SIP Phones. I had similar problem, when Asterisk
could not recognize my DTMF tones, so I had to tune the FXO modules. Here is
the link to the page:
http://www.voip-info.org/wiki/view/Asterisk+fxotune
If you are you using pure SIP Protocal, you may want to ask your SIP provider
for the suggested dtmfmode, even though RFC2833 is recommended by most. I did
have the same problems with this in the past, when I was testing with SIP
providers and I never solved it. Therefore, I went with the TDM Wildcard route
with analog lines. Good luck!!!!
Date: Thu, 16 Oct 2008 13:28:54 +0300
From: [EMAIL PROTECTED]
To: [email protected]
Subject: [asterisk-users] DTMF issue
Dear All,
I have the following scenario:
My customer dial a DID number and it'll be forwarded to my asterisk server by
the below trunk defined in sip.conf:
[sip_proxy1]
type=peer
context=stations
host=81.201.82.112
disallow=all
allow=g729
allow=alaw
allow=ulaw
dtmfmode=RFC2833
relaxdtmf=yes
canreinvite=no
The above trunk will use the context stations defined in extensions.conf as
follow:
[stations]
exten => _X.,1,Gotoif($[${EXTEN} = 112] ? 21)
exten => _X.,2,DeadAGI,a2billing.php|3
exten => _X.,3,Wait,2
exten => _X.,4,Hangup
exten => _X.,21,AGI,a2billing.php|3
exten => _X.,22,Hangup
The System will ask the user to enter his PIN number...The problem is that
sometime the system recognize the PIN entered and sometimes the PIN is not
recognized...I'm using RFC2833 as dmf mode...
I would like to know if my config is correct or I need o add something to it or
there is a BUG on asterisk server regarding DTMF?
Please note that I'm using Asterisk 1.4.21.2
Regards
_________________________________________________________________
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