Hi All, Does someone used net2phone with freeswitch. I am having no audio problem, below is my sip trace and sofia setting. Please let me know did I miss something? SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.178.239.32;branch=z9hG4bKZ38e140yHZSye;received=208.86.251.68;rport=5060 From: "Prod" <sip:Prod@10.178.239.32>;tag=epH2avec6ZXmm To: <sip:99901134346556@66.33.147.149
;tag=sbcsipuas_1_C50986_20140501113400982_b12sb08 Call-ID: ebeedfce-4be8-1232-63b0-00123f20329d Contact: <sip:66.33.147.149:5060> CSeq: 59143169 INVITE Server: sbc_3 Content-Type: application/sdp Content-Length: 152
v=0 o=SBCSIPUAS 1696166055 1 IN IP4 10.50.2.40 s=SBCSIPUAS SIP STACK v1.0 c=IN IP4 10.50.2.40 t=0 0 m=audio 23194 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ---------------------------------------------------------- aggressive-nat-detection [true] all-reg-options-ping [true] apply-inbound-acl [default] auth-calls [true] context [Inbound_Route] debug [info] dialplan [XML] disable-transcoding [true] disable-transfer [true] dtmf-duration [2000] enable-100rel [false] enable-timer [false] force-subscription-expires [60] ignore-183nosdp [no] inbound-bypass-media [true] inbound-codec-negotiation [generous] inbound-codec-prefs [G722,PCMU,PCMA,GSM] inbound-proxy-media [false] inbound-reg-force-matching-username [true] liberal-dtmf [true] manual-redirect [false] minimum-session-expires [60] outbound-codec-prefs [G722,PCMU,PCMA,GSM] registration-thread-frequency [30] rtp-end-port [42000] rtp-ip [10.178.239.32] rtp-start-port [ 16000] rtp-timeout-sec [60] session-timeout [60] sip-ip [10.178.239.32] sip-port [5060] sip-trace [false] tls [false] unregister-on-options-fail [true]