<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I do not use Gizmo for inbound, only out. I have a register line that looks like yours. In addition I have this:<div><br></div><div><div><div>[general]</div><div>context=nonesaid </div><div>allowguest=no</div><div>allowoverlap=yes</div><div>allowtransfer=yes</div><div>realm=<my system's host name></div><div>bindport=5060</div><div>bindaddr=0.0.0.0</div><div>srvlookup=yes</div><div>maxexpiry=3600</div><div>minexpiry=60</div><div>defaultexpiry=1200</div><div>qualifyfreq=60</div><div>notifymimetype=text/plain</div><div>disallow=all</div><div>allow=gsm</div><div>allow=ulaw</div><div>allow=alaw</div><div>mohinterpret=default</div><div>mohsuggest=default</div><div>language=en</div><div>videosupport=yes</div><div>callevents=yes</div><div>alwaysauthreject=yes</div><div>externip=<mypublicIP></div><div>localnet=192.168.0.0/255.255.255.0</div><div>rtptimeout=60</div><div>rtpholdtimeout=300</div><div>rtpkeepalive=60</div><div>allowsubscribe=yes</div><div>callcounter=yes</div><div>counteronpeer=yes</div><div>registertimeout=20</div><div>registerattempts=10</div><div>nat=yes</div><div>canreinvite=nonat</div><div><br></div></div></div><div><div>[gizmo5]</div><div>type=peer</div><div>host=198.65.166.131</div><div>fromdomain=proxy01.sipphone.com</div><div>canreinvite=no</div><div>dtmfmode=rfc2833</div><div>insecure=port,invite</div><div>qualify=yes</div><div>fromuser=<myusername></div><div>authuser=<myusername></div><div>defaultuser=<myusername></div><div>secret=<mypass></div><div><div>context=sip-out</div><div>disallow=all</div><div>allow=ulaw</div><div>allow=alaw</div><div><br></div><div><br></div><div>Then in extensions.conf I have this:</div><div><br></div><div><div>exten => _9XX.,1,SIPAddHeader(No-Answer: true)</div></div><div><div>exten => _9XX.,n,Dial(SIP/gizmo5/${EXTEN:1},20)</div><div><br></div><div><br></div><div><br></div><div><div><span class="Apple-style-span" style="font-family: Monaco; font-size: 10px; ">-- </span></div></div></div></div><div apple-content-edited="true"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; ">Jim Dickenson</div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; "><a href="mailto:dickenson@cfmc.com">mailto:dickenson@cfmc.com</a></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; min-height: 14px; "><br></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; ">CfMC</div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Monaco; "><a href="http://www.cfmc.com/">http://www.cfmc.com/</a></div><div><font class="Apple-style-span" face="Monaco" size="2"><span class="Apple-style-span" style="font-size: 10px; "><br></span></font></div></div></div><br class="Apple-interchange-newline"> </div><br><div><div>On Aug 5, 2009, at 6:02 AM, Rob wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>Yes ... as a matter of fact here is the sip.conf ... obviously private info removed ....</div> <div>[general]<br>register => <a href="mailto:1747xxxxxxx%3Axxxxxxxxxxxx@proxy01.sipphone.com">1747xxxxxxx:xxxxxxxxxxxx@proxy01.sipphone.com</a><br>port = 5060<br>bindaddr = 192.168.22.5<br>context = incoming<br>svrlookup=yes<br> ;dtmfmode=inband<br>allow=all<br>externip=76.98.xxx.xxx<br>localnet=<a href="http://192.168.22.0/255.255.255.0">192.168.22.0/255.255.255.0</a><br></div> <div> </div> <div>[<a href="http://proxy01.sipphone.com">proxy01.sipphone.com</a>]<br>nat=yes<br>;type=peer<br>type=friend<br>context=incoming<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=ilbc<br>dtmfmode=rfc2833<br>host=<a href="http://proxy01.sipphone.com">proxy01.sipphone.com</a><br> fromdomain=<a href="http://proxy01.sipphone.com">proxy01.sipphone.com</a><br>;insecure=very deprecated; use insecure=port,invite instead<br>insecure=port,invite<br>qualify=yes<br>secret=XXXXXXXXXXXX<br>authuser=1747XXXXXXX<br> fromuser=1747XXXXXXX<br>username=1747XXXXXXX<br>canreinvite=no<br><br><br></div> <div class="gmail_quote">On Wed, Aug 5, 2009 at 6:07 AM, Administrator TOOTAI <span dir="ltr"><<a href="mailto:admin@tootai.net">admin@tootai.net</a>></span> wrote:<br> <blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Rob a écrit :<br>> Hi all,<br>><br>Hi<br> <div class="im">> I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a<br>> while and it works fine .... I just added CALL OUT ... I have no problem<br>> with call setup ... the called party hears me ... but I can't hear them ....<br> > again if the call comes INTO the server both sides work fine.<br>><br>><br></div>Looks like a nat issue: do you have nat=yes and canreinvite=no in your<br>sip.conf for Gizmo5?<br><br>--<br>Daniel<br><br>_______________________________________________<br> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br><br>AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>Register Now: <a href="http://www.astricon.net/" target="_blank">http://www.astricon.net</a><br> <br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> </blockquote></div><br> _______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>Register Now: <a href="http://www.astricon.net">http://www.astricon.net</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br></div></body></html>