Does anyone have experience using ShoreTel SIP trunks to integrate an Asterisk system? I am having trouble when the ShoreTel system transfers an incoming call from a SIP trunk to the voicemail system. From the SIP traffic, it looks like it negotiates a codec correctly, but once the RTP stream starts the call drops or there is no audio. I see errors in Asterisk such as: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission xxxxxxxxxxxxxxxxxxxxxx at 192.168.x.x for seqno 104 (Critical Request) Has anyone run into this before or have any ideas? Thanks, Joe