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 email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com 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.
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@gmail.com> To: freeswitch-users@lists.freeswitch.org 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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org