sure

tcpdump to a file (tcpdump -w save.cap etc...)
and then put the file somewhere where we can access it

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




Le 29/06/2010 à 11:41, Syed Hussain a écrit :


I have tried the options below with no luck.

will tcpdump output suffice ?

please advise

thanks

S

Re: [Freeswitch-users] IP SIP Trunk errors/issues

Monday, June 28, 2010 8:12 AM
hi
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@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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