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
