Hi All Wondering if someone could help me out with a ptime issue I'm having? Some background info, we have a SIP client that uses G729@20ms for Wi-Fi, and G729@50ms for 3G. The 3G call is the one I'm having trouble with. When a call is made from the SIP client to the PSTN via our carrier when a ptime of 50ms is used, the call fails. The carrier does support G729 with variable ptimes. I also think that the audio stream sent back from the carrier is G729 50ms, I have confirmed this by analyzing the RTP stream in Wireshark. In SIP terms, Freeswitch receives a 183 with SDP from the carrier which gets passed to the SIP client. Freeswitch then gets a 200 OK from the carrier, but Freeswitch does not send an ACK to the carrier for this. When 20ms is used, the ACK is sent. This could just be a by-product of a different problem, but I've yet to pin-point what that is. In my vars.xml, I have both global and outbound codecs set to "=G729@50i,G729", I've also tried them the other way around but I don't think this makes a difference. I have late negotiation set to false on my internal profile, and disable transcoding set to true so that whatever is dynamically used on the A-leg, gets used on the B-leg. I'm not using absolute codec strings for G729@50i, as I would think this make things fairly static, and then affects Wi-Fi calls that are trying to use 20ms. Out of curiosity, i set the absolute codec string to 50ms ( {absolute_codec_string=G729@50i} within bridge application) and the same thing happened. Oh and also I have my codec negotiation to "generous". In terms of the call failing, what happens is that the PSTN device rings, the SIP client goes to a ringing state, but when the PSTN device is picked up the SIP client does not progress to the next step. I have noticed that the signalling by this point is still stuck on sending an ACK to the carrier, so signalling probably isn't working anyway. Finally, I have mod_com_g729 loaded with a couple of licenses on the server. The idea is to not use a license for this kind of call, as both legs are supposed to be using G729@50i. I believe this is happening too when making a call, because I can't see and encoder/decoder being created in the FS console. Has anyone had G729 with ptime of 50ms working in passthru mode when using mod_com_g729? I My Freeswitch version is about a year old now, but I would like to avoid updating if possible. I can provide console/SIP logs if required! I really am stuck on this one, so thanks in advance for any help! If this turns out to be beyond the scope of this mailing list, then Freeswitch consultation would not be out of the question.