[Freeswitch-users] Issue with Invites without SDP

[SIC FS LIST]

Steve,

Thanks for the assistance.  The entire console log after setting console
loglevel 9 is as follows:

2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382
f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in
CS_NEW!

Everything else is just the SIP messaging.

It's like FS never gets the call from sofia for some reason.  I tried the
following (your comments around the media make sense for sure):
-- remove the disable transcoding (this did not work)
-- remove the other codec parameters (this did not work)

To clarify what the box does:
-- takes an invite
-- responds with a 302 redirect with a modified contact header that looks
like $dn;npdi=yes;rn=$rn (if there is an rn)@$host info.  This is done with
using mod_xml_curl to provide the dynamic dialplan.

There are not ever any b legs.  Every call is an invite, 100 trying, 302,
ACK.  In this case we get an invite, a 100 trying and that's it.  Doing an
ngrep -d any -qW byline port 80 shows that there is not a xml_curl req and
on the console at loglevel 9 the only message from FS is this:

2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382
f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in
CS_NEW!

Sofia spits out all of the SIP messaging but that's it.  It looks nothing
like a normal call when I send the SDP.  It's a little odd.

Here is the modified profile after commenting out all of the info:
<profile name="external">
<aliases>
<alias name="outbound"/>
<alias name="nat"/> <!-- for backwards compatibility -->
</aliases>

<domains>
<domain name="all" alias="false" parse="true"/>
</domains>

<settings>
<param name="user-agent-string" value="lnpdal0001.sipinterchange.com"/>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="context" value="public"/>
<param name="enable-100rel" value="false"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="4.71.122.205"/>
<param name="sip-ip" value="4.71.122.205"/>
<param name="manage-presence" value="false"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="bind-params" value="transport=udp"/>
<param name="tls" value="$${external_ssl_enable}"/>
<!-- <param name="pass-rfc2833" value="true"/> -->
<!-- <param name="inbound-proxy-media" value="true"/> -->
<!-- <param name="inbound-bypass-media" value="true"/> -->
<!-- <param name="inbound-late-negotiation" value="true"/> -->
<param name="accept-blind-reg" value="false"/>
<param name="accept-blind-auth" value="true"/>
<param name="nonce-ttl" value="60"/>
<!-- <param name="disable-transcoding" value="true"/> -->
<param name="auth-calls" value="false"/>
<param name="inbound-reg-force-matching-username" value="false"/>
<param name="auth-all-packets" value="false"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<param name="challenge-realm" value="auto_to"/>
<param name="enable-3pcc" value="true"/>
</settings>
</profile>

Thanks again for the help.

SDR
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