[Freeswitch-users] IP SIP Trunk errors/issues

[Syed Hussain]

I have tried the options below with no luck.

David  will tcpdump output suffice ?

please advise

thanks

S
Re: [Freeswitch-users] IP SIP Trunk errors/issues
        	Monday, June 28, 2010 8:12 AM
        	
            
            
            From: 
            "Zuhair Raza" <engineerzuhairraza at gmail.com>
            	
            	
            	
        	To: 
        	freeswitch-users at lists.freeswitch.orghi
try this one 

 <condition field="network_addr" expression="^x\.x\.x\.x$">
     <anti-action application="reject" data="407"/>



On Mon, Jun 28, 2010 at 4:16 PM, Syed Hussain <uzairsh at yahoo.com> wrote:

Hi ,



I have subscribed to sonovoip a sip trunk provider who uses IP based authentication for SIP services ,



I am trying to make outbound international calls which are being
rejected by the provider for some reason, I am not able to figure out
if there is something wrong with the configuration.



I am listing out my configuration. Any help  would be greatly appreciated .

I have also tried #wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration from freeswitch wiki gives errors .




The voip configuration works perfectly with asterisk 1.6



My configuration is follows



Under

/usr/local/freeswitch/conf/sip_profiles/external



cat sonovoip.xml

<include>

  <gateway name="sonovoip">

    <param name="username" value="anything"/>

    <param name="password" value="anything"/>

    <param name="realm" value="x.x.x.x"/> <!--sonovoip ip -->

    <param name="register" value="false"/>

    <param name="caller-id-in-from" value="true"/>

        <param name="extension-in-contact" value="true"/>

  </gateway>

</include>





My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml



<extension name="sonovoip" continue="true">

      <condition field="destination_number" expression="^(00.*)$">

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

      </condition>

    </extension>



I did change the vars.xml  ports



  <!-- Internal SIP Profile -->

  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>

  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/>

  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5081"/>

  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>

  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- External SIP Profile -->

  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>

  <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>

  <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>

  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>

  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>



The debug



#pastebin.com/4MFhSWm9





Please advise if any other information is required.



Thanks



S



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-- 
Regards,
Zuhair Raza
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