[Freeswitch-users] Invalid profile

Jair Santos jairds at shaw.ca
Fri Jul 11 11:44:24 PDT 2008


>From the Xlite log I got this
 
[08-07-11]11:40:02.602 | Info | DNS | "DNS [query:24.67.78.200:5090]" |
sua::CDNS::StartTopologyLookup
[08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._udp.'
'24.67.78.200:5090' [default-port:5060][fallback to A/QA:T] - Results:" |
sua::CDNS::InternalShutdown
[08-07-11]11:40:02.602 | Info | DNS | "    0 -
[ip:24.67.78.200:5090][ttl:4294967295][P:0][W:0][query-name:24.67.78.200:509
0]" | sua::CDNS::InternalShutdown
[08-07-11]11:40:02.602 | Info | DNS | "DNS [query:24.67.78.200:5090]" |
sua::CDNS::StartTopologyLookup
[08-07-11]11:40:02.602 | Info | DNS | "DNS-Lookup performed: '_sip._tcp.'
'24.67.78.200:5090' [default-port:5060][fallback to A/QA:T] - Results:" |
sua::CDNS::InternalShutdown
[08-07-11]11:40:02.602 | Info | DNS | "    0 -
[ip:24.67.78.200:5090][ttl:4294967295][P:0][W:0][query-name:24.67.78.200:509
0]" | sua::CDNS::InternalShutdown
[08-07-11]11:40:02.602 | Debug | AbstractPhone |
"[1001 at 24.67.78.200:5090]::OnDNSLookupPlusDiscoveryProcessResult" |
AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult
[08-07-11]11:40:02.602 | Info | AbstractPhone | "Receiving notification
about firewall IP address: 0.0.0.0, voip always possible: 0" |
AbPhone::CAbstractPhone::OnFirewallInformation
[08-07-11]11:40:02.602 | Debug | AbstractPhone | "eOldFirewallType !=
eInFWType" |
AbPhone::CAccount::CCCMSink::OnDNSLookupPlusDiscoveryProcessResult
[08-07-11]11:40:02.602 | Info | GUI | "Receiving notification about firewall
IP address: 0.0.0.0, voip always possible: 0" |
eyeBeam::CPhoneFrame::OnNetworkFirewallInformation
 
 
 

 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jair
Santos
Sent: Friday, July 11, 2008 11:35 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Invalid profile


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: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, July 11, 2008 11:30 AM
To: freeswitch-users at lists.freeswitch.org
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: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Friday, July 11, 2008 11:17 AM
To: freeswitch-users at lists.freeswitch.org
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:mod_sofia at 24.67.78.200:5090
BIND-URL        sip:mod_sofia at 24.67.78.200:5090;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            1001 at 24.67.78.200
Contact         "FreeSwitch"
<sip:1001 at 192.168.1.1:23740;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: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jair
Santos
Sent: Friday, July 11, 2008 9:25 AM
To: freeswitch-users at lists.freeswitch.org
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:brian at freeswitch.org




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