[asterisk-users] RTP no sound on asterisk

[Jerry Geis]

Hi all, I seem to only be getting (1) call to sip_write() in 
channels/chan_sip.c

I have a very simple setup. one server (no cards) 2 polycom IP 330 phones.
Server is 192.168.1.150 and phone is DHCP. Nothing else on the network.
No firewall is enabled.

I call into the dialplan with:

exten => 112,1,Answer
exten => 112,n,Playback(demo-congrats)
exten => 112,n,Hangup

I see this executing on the CLI. However  I have no audio.

Enabling RTP debug I see the Got RTP packet but there are no send RTP 
packets going out.

I edited the source and put logging messages first in main/rtp.c and I 
saw the ast_rtp_raw_write() getting called 1 time.
so I backed up the tree. Got into channels/chan_sip.c sip_write() and it 
only gets called 1 time.

I have had a couple of times where I heard audio. Hangup up and tried 
again. And NO audio for bunch more times...

What can be causing my RTP issue and no audio?

Jerry