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[fdhege at gmail.com]

Hello,

I am trying to get a new freeswitch installation working and I have it
registering to a sip provider just fine and can receive inbound calls
but when I try and place a call out through the gateway the switch is
rejecting it due to the domain in the to of the INVITE.

Here is the INVITE

15:15:19.467351 IP (tos 0x0, ttl 64, id 40045, offset 0, flags
[none], proto: UDP (17), length: 1398) 207.4.223.35.5080 >
207.16.137.36.sip: SIP, length: 1370
INVITE sip:16173800299@207.16.137.36 SIP/2.0
Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bKcc3a14Zcg8NZp
Max-Forwards: 69
From: "FreeSWITCH" <sip:
6172384723@172.16.12.100;transport=udp>;tag=54FXray2SNaKN
To: <sip:18023800299@207.16.137.36>
Call-ID: 6956c6eb-b6a7-122c-5fb1-725bd701687a
CSeq: 114786724 INVITE

Here is the REGISTER for the same gateway

15:24:27.746984 IP (tos 0x0, ttl 64, id 40075, offset 0, flags
[none], proto: UDP (17), length: 661) 207.4.223.35.5080 >
207.16.137.36.sip: SIP, length: 633
REGISTER sip:207.16.137.36;transport=udp SIP/2.0
Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bK9H0SmN8jvavXp
Max-Forwards: 70
From: <sip: 6172384723@172.16.12.100;transport=udp>;tag=3jXBNmvUZ3XDe
To: <sip: 6172384723@172.16.12.100;transport=udp>

Here is my gateway config

<gateway name="voip1">
<param name="username" value="6172384723" />
<param name="password" value="testme" />
<param name="realm" value="172.16.12.100" />
<param name="proxy" value="207.16.137.36"/>
<param name="register-proxy" value="207.16.137.36"/>
<param name="register" value="true" />
<param name="stun-enabled" value="false"/>
<param name="from-domain" value="172.16.12.100"/>
<param name="aggressive-nat-detection" value="false"/>
</gateway>

Here is the bridge from the dialplan

<action application="bridge" data="sofia/gateway/voip1/1$1"/>

I'm currently running FreeSWITCH Version 1.0.trunk (13226M).

I hunted around trying to find another option in for the gateway
config that would set the to domain to the realm and still send the
packet to proxy but could not find one.

Any help/hints are welcome.

Thanks,

-Dale

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[msc at freeswitch.org]

What does your dialplan entry look like? Also, a debug trace and even a sip trace would be useful. You can use pastepin.freeswitch.org to paste a lot of stuff in a place where everyone can view it and not overwhelm the email list.

-MC

On Fri, May 8, 2009 at 12:36 PM, Dale <fdhege@gmail.com (fdhege@gmail.com)> wrote:

[fdhege at gmail.com]

Thanks for the response. Of course after sending the email I find the problem. The account was not properly setup on the other end to place outbound calls. So everything is working now.


Thanks,


-Dale

On May 8, 2009, at 6:45 PM, Michael Collins wrote: