Hi All<div>I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound provider. My Inbound calls route properly to extension 1000, however, the caller does not hear the extension ringing. The extension at 1000 does ring however and once I pick up the call, the conversation proceeds normally. I just can't figure out why the caller hears silence during phone ringing. I have tried putting instant_ringback in dialplan/public.xml:</div>
<div><br></div><div><div><extension name="vitel-inbound"></div><div> <condition field="destination_number" expression=""></div><div> <action application="set" data="instant_ringback=true"/></div>
<div> <action application="transfer" data="1000 XML default"/></div><div> </condition></div><div> </extension></div></div><div><br></div><div>The log lines during inbound calls show:</div>
<div><div>2010-06-02 07:55:15.913827 [NOTICE] mod_sofia.c:1992 Pre-Answer sofia/external/<a href="mailto:7816409134@64.2.142.15">7816409134@64.2.142.15</a>!</div><div>2010-06-02 07:55:15.916855 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/<a href="http://sip:1000@172.168.128.193:5060">sip:1000@172.168.128.193:5060</a> [BREAK]</div>
<div>2010-06-02 07:55:15.916855 [DEBUG] sofia.c:4167 Channel sofia/external/<a href="mailto:2123135987@64.2.142.15">2123135987@64.2.142.15</a> skipping state [early][183]</div><div>2010-06-02 07:55:15.916855 [DEBUG] switch_core_session.c:642 Send signal sofia/external/<a href="mailto:2123135987@64.2.142.15">2123135987@64.2.142.15</a> [BREAK]</div>
<div>2010-06-02 07:55:15.916855 [DEBUG] switch_ivr_originate.c:1124 Raw Codec Activation Success L16@8000hz 1 channel 20ms</div><div>2010-06-02 07:55:15.916855 [DEBUG] switch_core_codec.c:122 sofia/external/<a href="mailto:2123135987@64.2.142.15">2123135987@64.2.142.15</a> Push codec L16:10</div>
<div>2010-06-02 07:55:15.916855 [DEBUG] switch_ivr_originate.c:1189 Play Ringback Tone [%(2000,4000,440.0,480.0)]</div><div>2010-06-02 07:55:15.922793 [DEBUG] sofia.c:4172 Channel sofia/internal/<a href="http://sip:1000@172.168.128.193:5060">sip:1000@172.168.128.193:5060</a> entering state [calling][0]</div>
<div>2010-06-02 07:55:15.981765 [DEBUG] sofia.c:5866 IP 64.2.142.15 Rejected by acl "domains". Falling back to Digest auth.</div><div>2010-06-02 07:55:15.984797 [WARNING] sofia_reg.c:1873 Can't find user [<a href="mailto:prav_24.90.81.72@172.168.128.203">prav_24.90.81.72@172.168.128.203</a>]</div>
<div>You must define a domain called '172.168.128.203' in your directory and add a user with the id="prav_24.90.81.72" attribute</div><div>and you must configure your device to use the proper domain in it's authentication credentials.</div>
<div>2010-06-02 07:55:15.984797 [WARNING] sofia_reg.c:1033 SIP auth failure (INVITE) on sofia profile 'internal' for [<a href="mailto:5087203445@24.90.81.72">5087203445@24.90.81.72</a>] from ip 64.2.142.15</div></div>
<div><br></div><div>where 5087203445 is my inbound DID. Can someone please help decipher what the warning means and why there might not be any early media ringback..</div><div><br></div><div>thanks </div><div>-Praveen</div>
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