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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:<br><br>http://www.voip-info.org/wiki/view/Asterisk+fxotune<br><br>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!!!!<br><br><hr>Date: Thu, 16 Oct 2008 13:28:54 +0300<br>From: michofr@gmail.com<br>To: asterisk-users@lists.digium.com<br>Subject: [asterisk-users] DTMF issue<br><br><div dir="ltr">Dear All,<br><br>I have the following scenario:<br>My customer dial a DID number and it'll be forwarded to my asterisk server by the below trunk defined in sip.conf:<br><br>[sip_proxy1] <br>type=peer <br>
context=stations<br>host=<a href="http://81.201.82.112" target="_blank">81.201.82.112</a><br>disallow=all<br>allow=g729<br>allow=alaw<br>allow=ulaw <br>dtmfmode=RFC2833 <br>relaxdtmf=yes<br>canreinvite=no<br><br>The above trunk will use the context stations defined in extensions.conf as follow:<br>
<br><br>[stations]<br>exten => _X.,1,Gotoif($[${EXTEN} = 112] ? 21)<br>exten => _X.,2,DeadAGI,a2billing.php|3<br>exten => _X.,3,Wait,2<br>exten => _X.,4,Hangup<br>exten => _X.,21,AGI,a2billing.php|3<br>exten => _X.,22,Hangup<br>
<br>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...<br><br>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?<br>
<br>Please note that I'm using Asterisk <a href="http://1.4.21.2" target="_blank">1.4.21.2</a><br>Regards <br></div>
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