<html>
<head>
<style>
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Verdana
}
</style>
</head>
<body class='hmmessage'>
i am not sure what you are talking about. i have extensions and my sip trunk config in that file. see below<br><br>[200]<br>deny=0.0.0.0/0.0.0.0<br>type=friend<br>secret=200<br>qualify=yes<br>port=5060<br>pickupgroup=<br>permit=0.0.0.0/0.0.0.0<br>nat=yes<br>mailbox=200@device<br>host=dynamic<br>dtmfmode=rfc2833<br>dial=SIP/200<br>context=from-internal<br>canreinvite=yes<br>callgroup=<br>callerid=device <200><br>accountcode=<br>call-limit=50<br><br><br>[BW-SIP-A]<br>disallow=all<br>canreinvite=yes<br>dtmfmode=rfc2833<br>host=x.x.x.x<br>outboundproxy=x.x.x.x<br>progressinbound=yes<br>qualify=300<br>type=peer<br>allow=ulaw<br><br>[BW-SIP-B]<br>disallow=all<br>canreinvite=yes<br>dtmfmode=rfc2833<br>host=x.x.x.x<br>outboundproxy=x.x.x.x<br>progressinbound=yes<br>qualify=300<br>type=peer<br>allow=ulaw<br><br>[from-bandwidth-A]<br>disallow=all<br>type=peer<br>reinvite=yes<br>port=5060<br>insecure=invite,port<br>host=x.x.x.x<br>fromdomain=x.x.x.x<br>dtmfmode=rfc2833<br>context=from-trunk<br>canreinvite=no<br>allow=ulaw<br>qualify=300<br><br>[from-bandwidth-B]<br>disallow=all<br>type=peer<br>reinvite=yes<br>port=5060<br>insecure=invite,port<br>host=x.x.x.x<br>fromdomain=x.x.x.x<br>dtmfmode=rfc2833<br>context=from-trunk<br>canreinvite=no<br>allow=ulaw<br>qualify=300<br><br><br><hr id="stopSpelling">Date: Fri, 14 Aug 2009 12:09:15 -0500<br>From: crt.rojas@gmail.com<br>To: asterisk-users@lists.digium.com<br>Subject: Re: [asterisk-users] no ring tone<br><br>Hello<br><br>One question<br><br>In sip.con or sip_additionals.conf, in freepbx, the context of your client do you put <br>nat = yes<br><br>externip = XXXX<br><br>You put your public ip.<br><br>Are you sure that?<br><br><br>
Regards<br><br><div class="EC_gmail_quote">On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>></span> wrote:<br><blockquote class="EC_gmail_quote" style="padding-left: 1ex;">
<div>
i changed it and still didn't ring. however it did ring on one call to a cell phone but it hasn't done it again.<br><br><hr>Date: Fri, 14 Aug 2009 09:39:33 -0500<br>From: <a href="mailto:crt.rojas@gmail.com">crt.rojas@gmail.com</a><div class="EC_im">
<br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br></div><div class="EC_im">Subject: Re: [asterisk-users] no ring tone<br><br></div>Hello,<br><br>I never use externhost<br>
<br>y use \<br><br>externip=public ip<br><br>And work fine<br><br><br>Regards<br><br><div><div class="EC_im">On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>></span> wrote:<br>
</div><blockquote style="padding-left: 1ex;">
<div><div class="EC_im">
how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring.<br><br> Edit sip_nat.conf for proper NAT:<br>localnet=<a href="http://192.168.1.0/255.255.255.0">192.168.1.0/255.255.255.0</a> externhost=<a href="http://pbx.DOMAIN.com">pbx.DOMAIN.com</a> (Set your external hostname name here)<br>
externrefresh=10<br>fromdomain=DOMAIN.com (Set your external domain name here)<br>nat=yes<br>qualify=yes<br>canreinvite=no<br><br><br></div> Add extra codecs to /etc/asterisk/sip_custom.conf<br>allow=gsm allow=h261<br>allow=h263<br>
allow=h263p<br>videosupport=yes<br><hr>Windows Live™: Keep your life in sync. <a href="http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009">Check it out.</a></div><div class="EC_im">
<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><br></div></blockquote></div><br><br><hr>Windows Live™: Keep your life in sync. <a href="http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009">Check it out.</a></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><br></blockquote></div><br><br /><hr />Get back to school stuff for them and cashback for you. <a href='http://www.bing.com/cashback?form=MSHYCB&publ=WLHMTAG&crea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1' target='_new'>Try Bing now.</a></body>
</html>