Do not understand, what these multiple interfaces that you say?

thanks.

2011/6/24 Tony Graziano <tgrazi...@myitdepartment.net>

> they claim to be running sipx on multiple interfaces. we all know this is
> not a good idea.
> On Jun 24, 2011 4:02 PM, "Yuri Kurkarewicz" <sipx...@kapten.com.br> wrote:
> > *Has anyone seen if this post is true?*
> > **
> > *http://qxip.net/mediawiki/index.php/SIPX_IPv6_HACK*<
> http://qxip.net/mediawiki/index.php/SIPX_IPv6_HACK>
>
> > **
> > *Has anyone seen if this post is true?
> >
> > Could indicate what are the security measures required in the server
> OpenUcand
> > Sipxecs?
> >
> >
> > Thank you.*
> > **
> >
> >
> > SIPX <http://qxip.net/mediawiki/index.php/SipXecs_Hacks>: *SIPX IPv6
>
> > HACK*(very experimental)
> >
> > Documentation on sipXecs support for IPv6 is somewhat confusing or
> pointing
> > at possible issues with many posts suggesting to disable it completely
> (?) -
> > No fancy transformations and routing hacks with IPv6 Day coming up? As
> usual
> > there's FreeSWITCH to save the day! Let's try route some IPv6 traffic to
> our
> > FS/SIPX instance and setup a dedicated profile/ip to handle the traffic.
> > Since we're running on the same host, we'll proxy media to sipX and have
> FS
> > perform all translations - since it's great at all it does - why not?
> >
> > NOTICE: This hack is nothing more than a work in
> > progress/unfinished/unsecured...
> > PRE-REQUISITES:
> >
> > - sipXecs 4.4.0 or higher
> > - IPv6 enabled network & IPv6 address at sipX host (or 6to4 tunnel)
> > - DNS IPv6 AAAA records for your SIP topology
> > - SIP clients supporting IPv6 *(Linphone forever!)*
> >
> > THE LOGIC:
> >
> > - Create SIPX/FreeSwitch/IPv6 Profile running on port 15080 (separate ip)
> >
> > - IPv6 calls routed to FS/IPv6 via additional SRV
> > - UA[6] > IPv6 > FS/IPv6 > FS/IPv4 > sipXecs > UA[4]
> >
> > *NOTE: of course you could as well use plain 5060 since we're on a
> different
> > interface, we prefer to introduce no confusion at this stage*
> >
> >
> > CHANGES:
> >
> > *FreeSwitch:*
> >
> > Create new profile: in the directory
> > /etc/sipxpbx/freeswitch/conf/sip_profiles/ipv6gw.xml:
> >
> > <profile name="ipv6gw">
> > <gateways>
> > </gateways>
> > <aliases>
> > </aliases>
> > <domains>
> > <domain name="all" alias="false" parse="true"/>
> > </domains>
> > <settings>
> > <param name="debug" value="0"/>
> > <param name="sip-trace" value="no"/>
> > <param name="rfc2833-pt" value="101"/>
> > <param name="sip-port" value="15080"/>
> > <param name="dialplan" value="XML"/>
> > <param name="context" value="ipv6gw"/>
> > <param name="dtmf-duration" value="100"/>
> > <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> > <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
> > <param name="hold-music" value="$${hold_music}"/>
> > <param name="rtp-timer-name" value="soft"/>
> > <param name="local-network-acl" value="localnet.auto"/>
> > <param name="manage-presence" value="false"/>
> > <param name="inbound-codec-negotiation" value="generous"/>
> > <param name="nonce-ttl" value="60"/>
> > <param name="auth-calls" value="false"/>
> > <param name="accept-blind-auth" value="true"/>
> > <param name="rtp-ip" value="YOUR_IPv6_ADDRESS"/>
> > <param name="sip-ip" value="YOUR_IPv6_ADDRESS"/>
> > <param name="ext-rtp-ip" value="YOUR_IPv6_ADDRESS"/>
> > <param name="ext-sip-ip" value="YOUR_IPv6_ADDRESS"/>
> > <param name="rtp-timeout-sec" value="300"/>
> > <param name="rtp-hold-timeout-sec" value="1800"/>
> > <param name="tls" value="$${external_ssl_enable}"/>
> > <param name="tls-bind-params" value="transport=tls"/>
> > <param name="tls-sip-port" value="$${external_tls_port}"/>
> > <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
> > <param name="tls-version" value="$${sip_tls_version}"/>
> > </settings>
> > </profile>
> >
> >
> > Create separate dialplan: in the directory
> > /etc/sipxpbx/freeswitch/conf/dialplan/ipv6gw.xml:
> >
> > <include>
> > <context name="ipv6gw">
> > <extension name="unloop">
> > <condition field="${unroll_loops}" expression="^true$"/>
> > <condition field="${sip_looped_call}" expression="^true$">
> > <action application="deflect" data="${destination_number}"/>
> > </condition>
> > </extension>
> > <extension name="outside_call" continue="true">
> > <condition>
> > <action application="set" data="outside_call=true"/>
> > </condition>
> > </extension>
> > <extension name="call_debug" continue="true">
> > <condition field="${call_debug}" expression="^true$" break="never">
> > <action application="info"/>
> > </condition>
> > </extension>
> > <condition field="destination_number" expression="^(\d+)$"/>
> > <action application="set"
> > data="effective_caller_id_number=${outbound_caller_id_number}"/>
> > <action application="set"
> > data="effective_caller_id_name=${outbound_caller_id_name}"/>
> > <action application="bridge"
> > data="sofia/your.host.net/$0...@your.host.net:5080"/>
> > </condition>
> > </extension>
> > <X-PRE-PROCESS cmd="include" data="ipv6gw/*.xml"/>
> > </context>
> > </include>
> >
> > or simply start with this barebone and add your requirements/rules later:
> >
> > <context name="ipv6gw">
> > <extension name="ipv6gw">
> > <condition>
> > <action application="set" data="proxy_media=true"/>
> > <action application="bridge"
> > data="sofia/host.net/${destination_number}@your.host.net<http://host.net/$%7bdestination_number...@your.host.net>
> "/>
> > </condition>
> > </extension>
> > </context>
> >
> >
> > Activate the new configuration and reload mod_sofia from your shell:
> >
> > /opt/freeswitch/bin/fs_cli -x "reloadxml"
> > /opt/freeswitch/bin/fs_cli -x "reload mod_sofia"
> >
> > TEST IT:
> >
> > - Setup a DNS AAAA entry for your host (ipv6.host.net)
> > - Start your IPv6 SIP Client (Linphone, I bet!)
> > - Dial sip:x...@ipv6.host.net:15080
> > - Enjoy your IPv6 to IPv4 Call
> >
> >
> > NEXT:
> >
> > - Making some sense of the above... but it works!
>
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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