Hi All We're quite new to Freeswitch and are in the process of migrating from OpenSer (as an SBC) to Freeswitch. Mostly all is working well, except an oddity on DTMF. Our scenario: Telco/SIP Provider A is passing us calls using DTMF inband. We have a freeswitch configured as a SBC using 2 sip profiles (telco and internal) to topology hide and manage distribution of calls to the PBX servers located behind the SBC. The freeswitch will be handling up to a few hundred calls so we're trying to keep it lightweight. Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling customer needs. An example inbound call profile looks like this: <extension name="Inbound 124356"> <condition field="destination_number" expression="^(123456)$"> <action application="pre_answer"/> <action application="start_dtmf" /> <action application="bridge" data="sofia<mailto:sofia/external/123456@INTERNAL.PBX.IP:5060%22/>/external/123456@INTERNAL.PBX.IP:5060"/<mailto:sofia/external/123456@INTERNAL.PBX.IP:5060"/>> </condition> </extension> Initially when calling into the platform IVR type applications runinng on our PBX boxes would not work (you could hear the DTMF but the platform did not recognise the tones). We have had to add the appliation start_dtmf in order for Freeswitch to pass the DTMF to the Asterisk PBX behind the SBC. Interestingly on our OpenSer platform we just proxied the media (rtpproxy) with inband DTMF from the Telco and our PBX boxes recognised the inband DTMF tones on the PBX platforms and IVR type applications just worked. However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? We have confirmed both legs are PCMA and when using start_dtmf the first second of the call is clipped. Kind Regards,