[Freeswitch-users] Gateway Authentication

Tue Sep 27 20:02:32 MSD 2011

Were you able to get this working? If not, get a console debug log with a
siptrace and drop it on pastebin.freeswitch.org. Be sure to use "FreeSWITCH
Log" as the syntax highlighting.

at fs_cli you need to do:

console loglevel debug
sofia global siptrace on

That will turn on all the debug stuff you need. From there make the call
attempt and capture the output, then drop on pastebin. Give us the pb URL in
this thread.

Thanks,
MC

On Tue, Sep 20, 2011 at 9:34 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  Hello, I'm trying to make the move from Asterix, but I'm running into
> some difficulties. I'm try to bridge a call using our gateway however it
> doesn't work. In wireshark I can see I'm getting an SIP
> 401 Unauthorized error with a WWW-Authenticate header, after FS send the
> INVITE message to the gateway. However FS doesnt seem to respond to the
> request for Authentication.  Asterix responds correctly however I cant seem
> to make FS to do the same. Any help would be appreciated
>
>
>  INVITE sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
> Via: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKB49SZQHrgaaKc
> Max-Forwards: 8
> From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
> To: <sip:+15618911806 at 4.55.35.60:5070>
> Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
> CSeq: 17931324 INVITE
> Contact: <sip:1-F2la9 at 207.67.30.226:5060;transport=udp;gw=level3>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 227
> X-FS-Support: update_display
> Remote-Party-ID: "LyonL" <sip:+14142211800 at 207.67.30.226:5060
> >;party=calling;screen=yes;privacy=off
>
>  v=0
> o=FreeSWITCH 1316516832 1316516833 IN IP4 10.126.200.6
> s=FreeSWITCH
> c=IN IP4 10.126.200.6
> t=0 0
> m=audio 17944 RTP/AVP 0 8 18 101 13
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
>
>  SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 207.67.30.226;received=207.67.30.226
> ;branch=z9hG4bKB49SZQHrgaaKc;rport=42534
> From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
> To: <sip:+15618911806 at 4.55.35.60:5070
> >;tag=SD6soqf99-1367649635-1316534779161
> Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
> CSeq: 17931324 INVITE
> WWW-Authenticate: DIGEST
> qop="auth",nonce="BroadWorksXgst2td09Tbihi2qBW",algorithm=MD5,realm="BroadWorks"
> Content-Length: 0
>
>     <include>
>      <extension name="4142211800">
>         <condition field="destination_number" expression="^(\+?1)?
> (4142211800)$">
>          <action application="set" data="effective_caller_id_name=LyonL"/>
>          <action application="set" data="effective_caller_id_number=
> +14142211800"/>
>          <action application="bridge" data="sofia/gateway/level3/
> +15618911806"/>
>        </condition>
>      </extension>
>    </include>
>
>   <include>
>    <gateway name="level3">
>      <param name="apply-inbound-acl" value="level3"/>
>      <param name="username" value="1-F2la9"/>
>      <param name="password" value="password"/>>
>      <param name="realm" value="BroadWorks"/>
>      <param name="proxy" value="4.55.35.60:5070"/>
>      <param name="from-domain" value="207.67.30.226:5060"/>
>      <param name="dtmf-type" value="rfc2833"/>
>      <param name="extension-in-contact" value="true"/>
>      <param name="caller-id-in-from" value="true"/>
>      <param name="register" value="false"/>
>    </gateway>
>  </include>
>
>   <profile name="external">
>    <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>    <!-- This profile is only for outbound registrations to providers -->
>    <gateways>
>      <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>    </gateways>
>
>    <aliases>
>      <!--
>      <alias name="outbound"/>
>      <alias name="nat"/>
>      -->
>    </aliases>
>
>    <domains>
>      <domain name="all" alias="false" parse="true"/>
>    </domains>
>
>    <settings>
>      <param name="debug" value="0"/>
>  <!-- If you want FreeSWITCH to shutdown if this profile fails to load,
> uncomment the next line. -->
>  <!-- <param name="shutdown-on-fail" value="true"/> -->
>      <param name="sip-trace" value="no"/>
>      <param name="sip-capture" value="no"/>
>      <param name="rfc2833-pt" value="101"/>
>      <param name="sip-port" value="$${external_sip_port}"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="public"/>
>      <param name="dtmf-duration" value="2000"/>
>      <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>      <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>      <param name="hold-music" value="$${hold_music}"/>
>      <param name="rtp-timer-name" value="soft"/>
>      <!--<param name="enable-100rel" value="true"/>-->
>      <!--<param name="disable-srv503" value="true"/>-->
>      <!-- This could be set to "passive" -->
>      <param name="local-network-acl" value="localnet.auto"/>
>      <param name="manage-presence" value="false"/>
>
>      <!-- used to share presence info across sofia profiles
>  manage-presence needs to be set to passive on this profile
>  if you want it to behave as if it were the internal profile
>  for presence.
>      -->
>      <!-- Name of the db to use for this profile -->
>      <!--<param name="dbname" value="share_presence"/>-->
>      <!--<param name="presence-hosts" value="$${domain}"/>-->
>      <!--<param name="force-register-domain" value="$${domain}"/>-->
>      <!--all inbound reg will stored in the db using this domain -->
>      <!--<param name="force-register-db-domain" value="$${domain}"/>-->
>      <!-- ************************************************* -->
>
>      <!--<param name="aggressive-nat-detection" value="true"/>-->
>      <param name="inbound-codec-negotiation" value="generous"/>
>      <param name="nonce-ttl" value="60"/>
>      <param name="auth-calls" value="false"/>
>      <!--<param name="challenge-realm" value="auto_from"/>-->
>      <param name="user-agent-string" value="FreeSWITCH"/>
>      <!--
>  DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
>      -->
>      <param name="rtp-ip" value="$${local_ip_v4}"/>
>      <param name="sip-ip" value="$${local_ip_v4}"/>
>      <param name="ext-rtp-ip" value="auto-nat"/>
>      <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>      <param name="rtp-timeout-sec" value="300"/>
>      <param name="rtp-hold-timeout-sec" value="1800"/>
>      <!--<param name="enable-3pcc" value="true"/>-->
>
>      <!-- TLS: disabled by default, set to "true" to enable -->
>      <param name="tls" value="$${external_ssl_enable}"/>
>      <!-- additional bind parameters for TLS -->
>      <param name="tls-bind-params" value="transport=tls"/>
>      <!-- Port to listen on for TLS requests. (5081 will be used if
> unspecified) -->
>      <param name="tls-sip-port" value="$${external_tls_port}"/>
>      <!-- Location of the agent.pem and cafile.pem ssl certificates
> (needed for TLS server) -->
>      <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
>      <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
> work with TLSv1 -->
>      <param name="tls-version" value="$${sip_tls_version}"/>
>
>    </settings>
>  </profile>
>
>
>
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