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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
