[Freeswitch-users] Sharing presence between FS boxes

[Chris B. Ware]

Hi,

On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358).

If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that
if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B?

By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_"
as I read on wiki.

Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared
between boxes, and even presence works sometimes, but is not stable. 

Here my internal sip profiles config:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="configuration" description="Various Configuration">
    <configuration name="sofia.conf" description="Sofia Endpoint">
      <profiles>
        <profile name="internal">
          <domains>
            <domain name="all" alias="true" parse="false"/>
          </domains>
          <settings>
            <param name="debug" value="0"/>
            <param name="sip-trace" value="no"/>
            <param name="sip-capture" value="no"/>
            <param name="watchdog-enabled" value="no"/>
            <param name="watchdog-step-timeout" value="30000"/>
            <param name="watchdog-event-timeout" value="30000"/>
            <param name="log-auth-failures" value="true"/>
            <param name="forward-unsolicited-mwi-notify" value="false"/>
            <param name="context" value="default"/>
            <param name="rfc2833-pt" value="101"/>
            <param name="sip-port" value="$${internal_sip_port}"/>
            <param name="dialplan" value="XML"/>
            <param name="dtmf-duration" value="2000"/>
            <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
            <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
            <param name="rtp-timer-name" value="soft"/>
            <param name="rtp-ip" value="$${local_ip_v4}"/>
            <param name="sip-ip" value="$${local_ip_v4}"/>
            <param name="hold-music" value="$${hold_music}"/>
            <param name="apply-nat-acl" value="nat.auto"/>
            <param name="apply-inbound-acl" value="domains"/>
            <param name="local-network-acl" value="localnet.auto"/>
            <param name="record-path" value="$${recordings_dir}"/>
            <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
            <param name="manage-presence" value="true"/>
            <param name="inbound-codec-negotiation" value="generous"/>
            <param name="tls" value="$${internal_ssl_enable}"/>
            <param name="tls-bind-params" value="transport=tls"/>
            <param name="tls-sip-port" value="$${internal_tls_port}"/>
            <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
            <param name="tls-version" value="$${sip_tls_version}"/>
            <param name="nonce-ttl" value="60"/>
            <param name="auth-calls" value="$${internal_auth_calls}"/>
            <param name="inbound-reg-force-matching-username" value="true"/>
            <param name="auth-all-packets" value="false"/>
            <param name="ext-rtp-ip" value="auto-nat"/>
            <param name="ext-sip-ip" value="auto-nat"/>
            <param name="rtp-timeout-sec" value="300"/>
            <param name="rtp-hold-timeout-sec" value="1800"/>
            <param name="presence-hosts" value="_DISABLED_"/>
            <param name="odbc-dsn" value="systemDB:freeswitch:pazzword"/>
            <param name="challenge-realm" value="auto_from"/>
            <param name="outbound-proxy" value="sip::5060"/>
          </settings>
        </profile>
      </profiles>
    </configuration>
  </section>
</document>


Can somebody help?
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