<div dir="ltr">Set the hostname parameter in switch.conf.xml of both freeswitch boxes to the same name</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Mar 17, 2014 at 4:29 AM, Chris B. Ware <span dir="ltr"><<a href="mailto:chrisbware@yahoo.it" target="_blank">chrisbware@yahoo.it</a>></span> wrote:<br>
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Anyone can help?</div><div>Should I add all my domains on presence_hosts?</div><div><br></div><div style="font-style:normal;font-size:13px;background-color:transparent;font-family:HelveticaNeue,'Helvetica Neue',Helvetica,Arial,'Lucida Grande',sans-serif">
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<div dir="ltr"> <font face="Arial"> Il Venerd́ 14 Marzo 2014 19:18, Chris B. Ware <<a href="mailto:chrisbware@yahoo.it" target="_blank">chrisbware@yahoo.it</a>> ha scritto:<br> </font> </div><div><div class="h5"> <div>
<div><div><div style="font-size:10pt;font-family:HelveticaNeue,'Helvetica Neue',Helvetica,Arial,'Lucida Grande',sans-serif"><div>Hi,</div><div><br></div><div>On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358).</div>
<div><br></div><div>If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that</div><div>if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B?</div>
<div><br></div><div>By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_"</div><div>as I read on wiki.</div><div><br></div><div>Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared</div>
<div>between boxes, and even presence works sometimes, but is not
stable. </div><div><br></div><div>Here my internal sip profiles
config:</div><div><br></div><div><?xml version="1.0" encoding="UTF-8" standalone="no"?></div><div><document type="freeswitch/xml"></div><div> <section name="configuration" description="Various Configuration"></div>
<div> <configuration name="sofia.conf" description="Sofia Endpoint"></div><div> <profiles></div><div> <profile name="internal"></div><div> <domains></div>
<div> <domain name="all" alias="true" parse="false"/></div><div> </domains></div><div> <settings></div><div> <param name="debug" value="0"/></div>
<div> <param name="sip-trace" value="no"/></div><div>
<param name="sip-capture" value="no"/></div><div> <param name="watchdog-enabled" value="no"/></div><div> <param name="watchdog-step-timeout" value="30000"/></div>
<div> <param name="watchdog-event-timeout" value="30000"/></div><div> <param name="log-auth-failures" value="true"/></div><div> <param name="forward-unsolicited-mwi-notify" value="false"/></div>
<div> <param name="context" value="default"/></div><div> <param name="rfc2833-pt" value="101"/></div><div> <param name="sip-port" value="$${internal_sip_port}"/></div>
<div>
<param name="dialplan" value="XML"/></div><div> <param name="dtmf-duration" value="2000"/></div><div> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/></div>
<div> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/></div><div> <param name="rtp-timer-name" value="soft"/></div><div> <param name="rtp-ip" value="$${local_ip_v4}"/></div>
<div> <param name="sip-ip" value="$${local_ip_v4}"/></div><div> <param name="hold-music" value="$${hold_music}"/></div><div> <param name="apply-nat-acl" value="nat.auto"/></div>
<div>
<param name="apply-inbound-acl" value="domains"/></div><div> <param name="local-network-acl" value="localnet.auto"/></div><div> <param name="record-path" value="$${recordings_dir}"/></div>
<div> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/></div><div> <param name="manage-presence" value="true"/></div>
<div> <param name="inbound-codec-negotiation" value="generous"/></div><div> <param name="tls" value="$${internal_ssl_enable}"/></div><div> <param name="tls-bind-params" value="transport=tls"/></div>
<div> <param
name="tls-sip-port" value="$${internal_tls_port}"/></div><div> <param name="tls-cert-dir" value="$${internal_ssl_dir}"/></div><div> <param name="tls-version" value="$${sip_tls_version}"/></div>
<div> <param name="nonce-ttl" value="60"/></div><div> <param name="auth-calls" value="$${internal_auth_calls}"/></div><div> <param name="inbound-reg-force-matching-username" value="true"/></div>
<div> <param name="auth-all-packets" value="false"/></div><div> <param name="ext-rtp-ip" value="auto-nat"/></div><div> <param name="ext-sip-ip" value="auto-nat"/></div>
<div>
<param name="rtp-timeout-sec" value="300"/></div><div> <param name="rtp-hold-timeout-sec" value="1800"/></div><div> <param name="presence-hosts" value="_DISABLED_"/></div>
<div> <param name="odbc-dsn" value="systemDB:freeswitch:pazzword"/></div><div> <param name="challenge-realm" value="auto_from"/></div><div> <param name="outbound-proxy" value="sip::5060"/></div>
<div> </settings></div><div> </profile></div><div> </profiles></div><div> </configuration></div><div> </section></div><div></document></div><div><br></div><div>
<br></div><div>Can somebody
help?</div></div></div></div><br><br></div> </div></div></div> </div> </div> </div></div><br>_________________________________________________________________________<br>
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