<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>
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