Hi I'm having problems with bridging sip calls in Freeswitch after the revision from February 27 "only flush on break when its a blocking situation part 1". I am using Freeswitch in a phone center application where I first call an agent, then call a lead and then bridge the two calls. If I use a version after Feb 27. it takes about 5 seconds after bridging before any sound gets through. Here is an anonymized version of my sofia profile: <profile name="external"> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> <!-- This profile is only for outbound registrations to providers --> <gateways> <gateway name="provider"> <!--/// account username *required* ///--> <param name="username" value="user"/> <!--/// auth realm: *optional* same as gateway name, if blank ///--> <param name="realm" value="provider IP"/> <!--/// username to use in from: *optional* same as username, if blank ///--> <!--<param name="from-user" value="cluecon"/>--> <!--/// domain to use in from: *optional* same as realm, if blank ///--> <param name="from-domain" value="FS host IP"/> <!--/// account password *required* ///--> <param name="password" value="Password"/> <!--/// extension for inbound calls: *optional* same as username, if blank ///--> <param name="extension" value="test"/> <!--/// proxy host: *optional* same as realm, if blank ///--> <!--<param name="proxy" value="asterlink.com"/>--> <!--/// send register to this proxy: *optional* same as proxy, if blank ///--> <!--<param name="register-proxy" value="mysbc.com"/>--> <!--/// expire in seconds: *optional* 3600, if blank ///--> <!--<param name="expire-seconds" value="60"/>--> <!--/// do not register ///--> <param name="register" value="false"/> <!-- which transport to use for register --> <!--<param name="register-transport" value="udp"/>--> <!--How many seconds before a retry when a failure or timeout occurs --> <!--<param name="retry-seconds" value="30"/>--> <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> <!--<param name="caller-id-in-from" value="false"/>--> <!--extra sip params to send in the contact--> <!--<param name="contact-params" value="tport=tcp"/>--> <!-- Put the extension in the contact --> <!--<param name="extension-in-contact" value="true"/>--> <!--send an options ping every x seconds, failure will unregister and/or mark it down--> <!--<param name="ping" value="25"/>--> <!--<param name="cid-type" value="rpid"/>--> </gateway> </gateways> <aliases> <!-- <alias name="outbound"/> <alias name="nat"/> --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. --> <!-- <param name="shutdown-on-fail" value="true"/> --> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <!-- RFC 5626 : Send reg-id and sip.instance --> <!--<param name="enable-rfc-5626" value="true"/> --> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${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="enable-100rel" value="true"/>--> <!--<param name="disable-srv503" value="true"/>--> <!-- This could be set to "passive" --> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <!-- used to share presence info across sofia profiles manage-presence needs to be set to passive on this profile if you want it to behave as if it were the internal profile for presence. --> <!-- Name of the db to use for this profile --> <!--<param name="dbname" value="share_presence"/>--> <!--<param name="presence-hosts" value="$${domain}"/>--> <!--<param name="force-register-domain" value="$${domain}"/>--> <!--all inbound reg will stored in the db using this domain --> <!--<param name="force-register-db-domain" value="$${domain}"/>--> <!-- ************************************************* --> <!--<param name="aggressive-nat-detection" value="true"/>--> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <!-- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS! --> <param name="rtp-ip" value="FS host IP"/> <param name="sip-ip" value="FS host IP"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <!--<param name="enable-3pcc" value="true"/>--> <param name="rtcp-audio-interval-msec" value="5000"/> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="$${external_ssl_enable}"/> <!-- Set to true to not bind on the normal sip-port but only on the TLS port --> <param name="tls-only" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) --> <param name="tls-sip-port" value="$${external_tls_port}"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) --> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files --> <param name="tls-passphrase" value=""/> <!-- Verify the date on TLS certificates --> <param name="tls-verify-date" value="true"/> <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate --> <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe --> <param name="tls-verify-policy" value="none"/> <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none --> <param name="tls-verify-depth" value="2"/> <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe --> <param name="tls-verify-in-subjects" value=""/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 --> <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile> -- Tor Petterson tpe@actimizer.com Tobaksvejen 25, 2. tv. - 2860 Søborg Telephone: +45 39 55 05 32 www.actimizer.com