[Freeswitch-users] Sharing presence between FS boxes
Chris B. Ware
chrisbware at yahoo.it
Mon Mar 17 11:29:34 MSK 2014
Anyone can help?
Should I add all my domains on presence_hosts?
Il Venerdì 14 Marzo 2014 19:18, Chris B. Ware <chrisbware at 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?
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