[Freeswitch-users] Freeswitch with Net2Phone

[Umair Khan]

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 at 10.178.239.32>;tag=epH2avec6ZXmm
   To: <sip:99901134346556 at 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]
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