[Freeswitch-users] Issue with Invites without SDP

[Steven Ayre]

"Everything else is just the SIP messaging."

And dialplan/call processing... that's precisely the context it's useful to see.

-Steve


On 10 November 2010 16:22, SIC FS LIST <sicfslist at gmail.com> wrote:
> 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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