Anyone can help?

Should I add all my domains on presence_hosts?

Il Venerd́ 14 Marzo 2014 19:18, Chris B. Ware <chrisbware@yahoo.it> ha scritto:

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?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org