thx a lot for your explanation, i have another question, my sip server (mera sip-hit) dont need user name and password to connect. for security i just use firewall rules.<div>if i refer to other samples like fwd,voicheap etc all require user name and password.</div>
<div><br><div>my conf/sip_profiles/external/test.xml look like this </div><div><div> <include></div><div> <gateway name="test"></div><div> <param name="realm" value="202.xx.xx.xx.xx"/></div>
<div> <param name="proxy" value="202.xx.xx.xx.xx"/></div><div> <param name="expire-seconds" value="3600"/></div><div> <param name="register" value="true"/></div>
<div> <param name="retry-seconds" value="3600"/></div><div> </gateway></div><div></include></div><div><br></div><div><br></div><div>my conf/dialplan/default/00_test.xml look like</div>
<div><br></div><div><div><include></div><div> <extension name="test"></div><div> <condition field="destination_number" expression="^(\d)$"></div><div> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/></div>
<div> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/></div><div> <action application="bridge" data="sofia/gateway/test/$1"/></div>
<div> </condition></div><div> </extension></div><div></include></div><div><br></div><div><br></div><div>but still i cant see any call coming from my mera siphit.</div><div><br></div><div>TIA</div><div>
<br></div><div>budi wibowo</div></div></div><div><br><div><br><br><div class="gmail_quote">On Sat, May 8, 2010 at 4:04 PM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com">david.ponzone@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">Not sure because your logs are toon short, but it sems that your call is routed to ENUM.<div>
In the default config, ENUM routing is the last resort rule.</div><div>You need to add your own stuff before it.</div><div>To do so, you may add your extensions in a xml file called 01_something.xml in conf/dialplan/default</div>
<div>(see 01_example.com.xml for an example).</div><div><br><div> <span style="border-collapse:separate;color:rgb(0, 0, 0);font-family:Helvetica;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div style="word-wrap:break-word">
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<span><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;text-align:justify;color:rgb(192, 192, 192)"><i>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. </i><b><i>IPeva</i></b><i> 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.</i></div>
<div style="text-decoration:underline;text-align:justify"><font color="#C0C0C0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br></div></span><br> </div><br><div><div>Le 08/05/2010 à 10:26, budi wibowo a écrit :</div>
<div><div></div><div class="h5"><br><blockquote type="cite">dear all i try to make some changes but my call still failing<div><div>why the call never reach gateway i define in sip_profile/external/ and call goes to <a href="http://voiprakyar.or.id" target="_blank">voiprakyar.or.id</a> that i never define </div>
<div><br></div><div><br></div><div><br></div><div><br></div><div>2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel sofia/internal/<a href="mailto:budi@sip1.xxx.com" target="_blank">budi@sip1.xxx.com</a> [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]</div>
<div>2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing budi->62815145150 in context default</div><div>2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/<a href="mailto:budi@sip1.xxx.com" target="_blank">budi@sip1.xxx.com</a> to enum[62815145150@default]</div>
<div>2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel sofia/internal/<a href="mailto:62815145150@voiprakyat.or.id" target="_blank">62815145150@voiprakyat.or.id</a> [1e55a5c7-5d58-4437-a48b-4278154b57a0]</div>
<div>2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/<a href="mailto:62815145150@voiprakyat.or.id" target="_blank">62815145150@voiprakyat.or.id</a> [CS_CONSUME_MEDIA] [CALL_REJECTED]</div> <div><br>
</div><div><br></div><div>TIA</div><div><br></div><div>budi</div><br><div class="gmail_quote">On Sat, May 8, 2010 at 9:30 AM, Seven Du <span dir="ltr"><<a href="mailto:dujinfang@gmail.com" target="_blank">dujinfang@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">In the mean time, I think it's a good article for you:<br> <br> <a href="http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT" target="_blank">http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT</a><br>
<br> 2010/5/8 Seven Du <<a href="mailto:dujinfang@gmail.com" target="_blank">dujinfang@gmail.com</a>>:<br> <div><div></div><div>> default configuration files of mod_sofia is in conf/sip_profiles<br> ><br> > <a href="http://wiki.freeswitch.org/wiki/Sofia" target="_blank">http://wiki.freeswitch.org/wiki/Sofia</a><br>
> <a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files</a><br> ><br> > 2010/5/8 budi wibowo <<a href="mailto:bwibowo@gmail.com" target="_blank">bwibowo@gmail.com</a>>:<br>
>> thx, after reading the document i still got confusion about sofia module<br> >> i give example below<br> >><br> >> Dialing Through A Gateway(SIP Provider)<br> >><br> >> A gateway is a means for making outbound calls through a SIP provider. For<br>
>> example:<br> >><br> >> sofia/gateway/<a href="http://mygateway.com/1234" target="_blank">mygateway.com/1234</a><br> >><br> >> by default i dont find any directory named sofia, where i should put this<br>
>> directory?<br> >> many document i read telling about sofia<br> >> TIA<br> >> budi<br> >><br> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>> wrote:<br>
>>><br> >>> <a href="http://wiki.freeswitch.org/wiki/Dialplan" target="_blank">http://wiki.freeswitch.org/wiki/Dialplan</a><br> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:<br> >>><br>
>>> Yes I have siphit installed, I tried to make some changes on dialplan file<br> >>> but call always goes to other server, what should I configure to implement<br> >>> this outgoing call<br> >>><br>
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