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<p class=MsoNormal>Hello, <o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I’m a little confused on DTMF.<o:p></o:p></p>
<p class=MsoNormal>A sip peer is registered on two Asterisk servers. No
dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk
2. They both register on each other.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>A call comes in on Asterisk server 1, provider 1, dtmf=inband.
Then the call is transferred to Asterisk 2:<o:p></o:p></p>
<p class=MsoNormal>RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351@65.xx.xx.10,,tT,)<o:p></o:p></p>
<p class=MsoNormal>Where 12351 accepts the call on Asterisk 2, and in some
cases, that call is transferred out to a PSTN number, or wherever, but not
within Asterisk anymore via provider2, dtmf=rfc2833.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>When the call comes in, I’d like it to relay DTMF just
dandy. How can I do so?<o:p></o:p></p>
<p class=MsoNormal>There is no NAT between the Asterisk servers or in front of
them. However, Asterisk2 has iptables which allows all UDP traffic to/fro
Asterisk1. When Asterisk2 transfers the call to external endpoints, there might
be a LAN, but relative ports are open on those LANs.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Please help.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Thanks in advance,<o:p></o:p></p>
<p class=MsoNormal>Mark.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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