<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif; "><div><span class="Apple-style-span" style="font-size: 12px; color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; ">I am facing delay problems with mod_dingaling. The voice delay increases each minute that I am on an outgoing call. It starts fine and then will get up to 5-10 seconds within 3 minutes. The delay only occurs on the voice of the called party. I have googled and read countless threads and entries. I am running freeswitch on a VM. I have run my setup on CentOS 5.6 i386, CentOS 5.6 x86_64, CentOS 6 x86_64. I have tried using a stun server as well as port forwarding from my router. I have tried using the rtp-autoflush=true and rtp-timer-name=none settings. All of my changes have given me the same results. I am using the current GIT version. I have seen a number of people have a flawless experience with this, so what am I doing differently? Any help?</span></div><span id="OLK_SRC_BODY_SECTION"><div><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif; "><div><div style="color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; font-size: 12px; background-color: rgb(255, 255, 255); "><br></div><div style="color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; font-size: 12px; background-color: rgb(255, 255, 255); ">Here are my current jingle configurations:</div><div style="color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; font-size: 12px; background-color: rgb(255, 255, 255); "><br></div><div style="color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; font-size: 12px; background-color: rgb(255, 255, 255); "><div><extension name="gvoice_out"></div><div> <condition field="destination_number" expression="^1(\d{10})$"></div><div> <action application="set" data="rtp_autoflush=true"/></div><div> <action application="set" data="hangup_after_bridge=true"/></div><div> <action application="ring_ready"/></div><div> <action application="bridge" data="<a href="mailto:dingaling/gtalk/+1$1@voice.google.com">dingaling/gtalk/+1$1@voice.google.com</a>"/></div><div> </condition></div><div> </extension></div></div><div style="color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; font-size: 12px; background-color: rgb(255, 255, 255); "><br></div><div style="color: rgb(69, 69, 69); font-family: Arial, Helvetica, sans-serif; font-size: 12px; background-color: rgb(255, 255, 255); "><div><profile type="client"></div><div> <param name="name" value="gtalk"/></div><div> <param name="login" value="<email>@gmail.com/talk"/></div><div> <param name="password" value="***"/></div><div> <param name="dialplan" value="XML"/></div><div> <param name="context" value="public"/></div><div> <param name="message" value="FreeSwitch for everybody"/></div><div> <param name="rtp-ip" value="auto"/></div><div> <param name="auto-login" value="true"/></div><div> <param name="sasl" value="plain"/></div><div> <param name="server" value="talk.google.com"/></div><div> <param name="tls" value="true"/></div><div> <param name="use-rtp-timer" value="true"/></div><div> <param name="exten" value="1000"/></div><div> <param name="vad" value="both"/></div><div> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--></div><div> <param name="candidate-acl" value="wan.auto"/></div><div> <param name="local-network-acl" value="localnet.auto"/></div><div> <param name="rtp-timer-name value="none"/></div><div> <param name="rtp-autoflush" value="true"/></div><div> </profile></div></div></div></div></div></span></body></html>