What does your dialplan entry look like? Also, a debug trace and even a sip trace would be useful. You can use <a href="http://pastepin.freeswitch.org">pastepin.freeswitch.org</a> to paste a lot of stuff in a place where everyone can view it and not overwhelm the email list.<br>
<br>-MC<br><br><div class="gmail_quote">On Fri, May 8, 2009 at 12:36 PM, Dale <span dir="ltr"><<a href="mailto:fdhege@gmail.com">fdhege@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
Hello,<br>
<br>
I am trying to get a new freeswitch installation working and I have it<br>
registering to a sip provider just fine and can receive inbound calls<br>
but when I try and place a call out through the gateway the switch is<br>
rejecting it due to the domain in the to of the INVITE.<br>
<br>
Here is the INVITE<br>
<br>
15:15:19.467351 IP (tos 0x0, ttl 64, id 40045, offset 0, flags<br>
[none], proto: UDP (17), length: 1398) 207.4.223.35.5080 ><br>
207.16.137.36.sip: SIP, length: 1370<br>
INVITE <a href="mailto:sip%3A16173800299@207.16.137.36">sip:16173800299@207.16.137.36</a> SIP/2.0<br>
Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bKcc3a14Zcg8NZp<br>
Max-Forwards: 69<br>
From: "FreeSWITCH" <sip:<br>
<a href="mailto:6172384723@172.16.12.100">6172384723@172.16.12.100</a>;transport=udp>;tag=54FXray2SNaKN<br>
To: <<a href="mailto:sip%3A18023800299@207.16.137.36">sip:18023800299@207.16.137.36</a>><br>
Call-ID: 6956c6eb-b6a7-122c-5fb1-725bd701687a<br>
CSeq: 114786724 INVITE<br>
<br>
Here is the REGISTER for the same gateway<br>
<br>
15:24:27.746984 IP (tos 0x0, ttl 64, id 40075, offset 0, flags<br>
[none], proto: UDP (17), length: 661) 207.4.223.35.5080 ><br>
207.16.137.36.sip: SIP, length: 633<br>
REGISTER sip:207.16.137.36;transport=udp SIP/2.0<br>
Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bK9H0SmN8jvavXp<br>
Max-Forwards: 70<br>
From: <sip: <a href="mailto:6172384723@172.16.12.100">6172384723@172.16.12.100</a>;transport=udp>;tag=3jXBNmvUZ3XDe<br>
To: <sip: <a href="mailto:6172384723@172.16.12.100">6172384723@172.16.12.100</a>;transport=udp><br>
<br>
Here is my gateway config<br>
<br>
<gateway name="voip1"><br>
<param name="username" value="6172384723" /><br>
<param name="password" value="testme" /><br>
<param name="realm" value="172.16.12.100" /><br>
<param name="proxy" value="207.16.137.36"/><br>
<param name="register-proxy" value="207.16.137.36"/><br>
<param name="register" value="true" /><br>
<param name="stun-enabled" value="false"/><br>
<param name="from-domain" value="172.16.12.100"/><br>
<param name="aggressive-nat-detection" value="false"/><br>
</gateway><br>
<br>
Here is the bridge from the dialplan<br>
<br>
<action application="bridge" data="sofia/gateway/voip1/1$1"/><br>
<br>
I'm currently running FreeSWITCH Version 1.0.trunk (13226M).<br>
<br>
I hunted around trying to find another option in for the gateway<br>
config that would set the to domain to the realm and still send the<br>
packet to proxy but could not find one.<br>
<br>
Any help/hints are welcome.<br>
<br>
Thanks,<br>
<br>
-Dale<br>
<br>
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</blockquote></div><br>