[asterisk-users] ShoreTel <-> Asterisk Integration

[Joe Evans]

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