[Freeswitch-users] Issue with Invites without SDP

[SIC FS LIST]

Hello,

I have a working FS instance I am using as a redirect server (to serve up
LNP requests).  It works fine on calls with invites that have SDP and does
not work with invites without SDP.  I enabled 3pcc to true thinking that
would fix the issue.  Version info is FreeSWITCH Version 1.0.6
(hacked-20100921T052029Z).

With the console log level set to debug the only thing I see is this message
(just before returning a 480):

freeswitch at lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT]
switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout
waiting for next instruction in CS_NEW!

I also do not ever see FS to an xml http req (which is how we control the
dialplan).

freeswitch at lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at
14:57:50.934431:
------------------------------------------------------------------------
INVITE sip:19033226103 at X.X.X.X:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP
X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5
Max-Forwards: 70
Contact: <sip:X.X.X.X:5060>
To: <sip:19033226103 at X.X.X.X:5060>
From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001
Call-ID: 993186-34928005397-80464 at ens.com
CSeq: 693501 INVITE
Date: Tue, 09 Nov 2010 15:29:25 GMT
Supported: 100rel
User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152
Content-Length: 0

------------------------------------------------------------------------
send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5
From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001
To: <sip:19033226103 at X.X.X.X:5060>
Call-ID: 993186-34928005397-80464 at ens.com
CSeq: 693501 INVITE
User-Agent: lnpdal0001.sipinterchange.com
Content-Length: 0

------------------------------------------------------------------------
send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429:
------------------------------------------------------------------------
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP
X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5
From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001
To: <sip:19033226103 at X.X.X.X:5060>;tag=t7ycB64mX33mQ
Call-ID: 993186-34928005397-80464 at ens.com
CSeq: 693501 INVITE
User-Agent: lnpdal0001.sipinterchange.com
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE"
Content-Length: 0
Remote-Party-ID: "19033226103" <sip:19033226103 at X.X.X.X
>;party=calling;privacy=off;screen=no

------------------------------------------------------------------------

Here is the sofia profile:

<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"/>
<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="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="X.X.X.X"/>
<param name="sip-ip" value="X.X.X.X"/>
<param name="manage-presence" value="false"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="bind-params" value="transport=udp"/>
<param name="tls" value="$${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>

Any thoughts / ideas / help would be greatly appreciated.

Thanks!

SDR
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