Make sure you have the RTP ports forwarded in on your router. From: app_rpt-users-bounces@ohnosec.org [mailto:app_rpt-users-bounces@ohnosec.org] On Behalf Of Scott Sent: Saturday, November 10, 2012 5:45 PM To: App_rpt-users@ohnosec.org Subject: [App_rpt-users] sip.allstarlink.org sip session no-audio When trying to connect from a sip-phone to the portal, sip.allstarlink.org using, sip://1:allstar@sip.allstarlink.org:5060/0 -or- sip://1:allstar@sip.allstarlink.org:5060/<node_number> -or- sip://0@sip.allstarlink.org It connects, but there's no audio. No errors. PCMU@8000 correctly negotiated. I've tried two SIP-phone clients and a FreeSWITCH user-agent with the same negative outcomes. I've tried variations on the connect stream with same negative results. The BRIA soft-phone even handshakes GSM but still no audio. Is there something I'm missing? Thanks,