[Freeswitch-users] Issue with Invites without SDP

[Steven Ayre]

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)

If you think sofia's not passing the call to FS you can check the
internal processing of the sofia stack using "sofia loglevel all 9".
That'll show it picking up the call and any errors within the stack
before it reaches FS.

-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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