sipx only operates properly with one interface (network card) at this time. On Jun 24, 2011 4:15 PM, "Yuri Kurkarewicz" <[email protected]> wrote: > Do not understand, what these multiple interfaces that you say? > > thanks. > > 2011/6/24 Tony Graziano <[email protected]> > >> 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" <[email protected]> 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/[email protected]: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/${destination_number}@your.host.net< http://host.net/$%[email protected]> >> "/> >> > </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:[email protected]:15080 >> > - Enjoy your IPv6 to IPv4 Call >> > >> > >> > NEXT: >> > >> > - Making some sense of the above... but it works! >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
