Where did you put it in the dialplan? At the bottom? You'll wanna put it before the transfer to enum otherwise you'll always get NO ROUTE

Location Location Location.. that's what they tell me.

/b

On Jul 11, 2008, at 1:35 PM, Jair Santos wrote:

I am getting "call failed not found" and a voice message saying "The person you are calling is not available".

It seems that this error is coming from the provider but the other phone (inside the NAT) can make the calls.

I can call the inside the NAT phone from outside.

I cannot call the outside phone from the one inside the NAT. I get a busy signal.
thanks

Jair

-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brian West
Sent: Friday, July 11, 2008 11:30 AM
To: [email protected]
Subject: Re: [Freeswitch-users] Invalid profile

That should work what kind of error are you seeing.....

/b

On Jul 11, 2008, at 1:24 PM, Jair Santos wrote:
I have the following dialplan defined. It works from a phone inside the network.

<extension name="teste">
   <condition field="destination_number" expression="^(\d+)$">
<!--- The % behind the username tells FS to lookup the user in it's local sip_registration database --> <action application="bridge" data="sofia/gateway/inphonex/ $0"/>
   </condition>
 </extension>

and the following gateway in the directory


<include>
   <user id="inphonex">
     <gateways>
        <gateway name="inphonex">
                <param name="username" value="3462101"/>
                <param name="realm" value="sip.varphonex.com"/>
                <param name="password" value="606546"/>
                <param name="proxy" value="sip.varphonex.com"/>
                <param name="register" value="true"/>
                <param name="expire-seconds" value="3600"/>
         </gateway>
</gateways>
<params>
       <param name="password" value="606546"/>
     </params>
   </user>
</include>


-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brian West
Sent: Friday, July 11, 2008 11:17 AM
To: [email protected]
Subject: Re: [Freeswitch-users] Invalid profile

How are you sending calls from your dialplan to your provider?

/b

On Jul 11, 2008, at 1:02 PM, Jair Santos wrote:
Regarding my previous email, I have now the profile registered but still cannot make PSTN calls from the Phone

sofia status profile doublenat
API CALL [sofia(status profile doublenat)] output:
= = = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================
Name            doublenat
Domain Name     doublenat
DBName          sofia_reg_doublenat
Dialplan        XML
RTP-IP          192.168.1.117
Ext-RTP-IP      stun:stun.freeswitch.org
SIP-IP          192.168.1.117
Ext-SIP-IP      24.67.78.200
URL             sip:[EMAIL PROTECTED]:5090
BIND-URL        sip:[EMAIL PROTECTED]:5090;maddr=192.168.1.117
HOLD-MUSIC      local_stream://moh
CODECS          PCMU,PCMA,GSM
TEL-EVENT       101
CNG             13
SESSION-TO      0
MAX-DIALOG      0

Registrations:
= = = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================
Call-ID         MGQ0YmQyYmMxNmNmYTkyYzA4MTVjNTNjMDFkYjM4YzI.
User            [EMAIL PROTECTED]
Contact "FreeSwitch" <sip:[EMAIL PROTECTED]:23740;rinstance=5ea23586706cd62a >;nat
Agent           X-Lite release 1100l stamp 47546
Status Registered(AUTO-NAT)(unknown) EXP(2008-07-11 12:44:22)

= = = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================


Jair Santos


-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Jair Santos
Sent: Friday, July 11, 2008 9:25 AM
To: [email protected]
Subject: [Freeswitch-users] Invalid profile

Hi,

I am trying to register a phone in the following scenario:
FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100)

I followed the instructions from http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and have created the profile like:

<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="external">
<!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <alias name="outbound"/>
  </aliases>

  <domains>
    <domain name="$${domain}" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="apply-nat-acl" value="rfc1918"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5090"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="manage-presence" value="false"/>
    <param name="aggressive-nat-detection" value="true"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-timeout-sec" value="1800"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>

In the phone I am using the {external IP} :5090. I went to the router and forwarded the 5090 to the machine where FS is running.

The phone register and I can call internal extensions but I cannot place calls to PSTN (Call Failed)

The profile is invalid.

  sofia status profile doublenat
API CALL [sofia(status profile doublenat)] output:
Invalid Profile!

Does anybody know what I am missing ?

thanks

Jair Santos













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Brian West
sip:[EMAIL PROTECTED]



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Brian West
sip:[EMAIL PROTECTED]



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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Brian West
sip:[EMAIL PROTECTED]



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