| Index: webrtc/voice_engine/channel.cc
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| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
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| index 7895e9b9e142a283c311730d953f03e06687c2db..582bde5f26216f645340ef2102d87e485ff2fafb 100644
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| --- a/webrtc/voice_engine/channel.cc
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| +++ b/webrtc/voice_engine/channel.cc
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| @@ -809,6 +809,7 @@
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|                                             this,
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|                                             this,
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|                                             rtp_payload_registry_.get())),
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| +      telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
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|        _outputAudioLevel(),
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|        _externalTransport(false),
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|        // Avoid conflict with other channels by adding 1024 - 1026,
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| @@ -978,6 +979,7 @@
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|    // disabled by the user.
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|    // After StopListen (when no sockets exists), RTCP packets will no longer
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|    // be transmitted since the Transport object will then be invalid.
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| +  telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
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|    // RTCP is enabled by default.
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|    _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
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|    // --- Register all permanent callbacks
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| 
 |