[Freeswitch-users] outgoing gateway

budi wibowo bwibowo at gmail.com
Sat May 8 17:16:15 PDT 2010


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.
if i refer to other samples like fwd,voicheap etc all require user name and
password.

my conf/sip_profiles/external/test.xml look like this
 <include>
        <gateway name="test">
          <param name="realm" value="202.xx.xx.xx.xx"/>
          <param name="proxy" value="202.xx.xx.xx.xx"/>
          <param name="expire-seconds" value="3600"/>
          <param name="register" value="true"/>
          <param name="retry-seconds" value="3600"/>
        </gateway>
</include>


my conf/dialplan/default/00_test.xml look like

<include>
   <extension name="test">
     <condition field="destination_number" expression="^(\d)$">
       <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
       <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
       <action application="bridge" data="sofia/gateway/test/$1"/>
     </condition>
   </extension>
</include>


but still i cant see any call coming from my mera siphit.

TIA

budi wibowo



On Sat, May 8, 2010 at 4:04 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Not sure because your logs are toon short, but it sems that your call is
> routed to ENUM.
> In the default config, ENUM routing is the last resort rule.
> You need to add your own stuff before it.
> To do so, you may add your extensions in a xml file called 01_something.xml
> in conf/dialplan/default
> (see 01_example.com.xml for an example).
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
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>
> Le 08/05/2010 à 10:26, budi wibowo a écrit :
>
> dear all i try to make some changes but my call still failing
> why the call never reach gateway i define in sip_profile/external/ and call
> goes to voiprakyar.or.id that i never define
>
>
>
>
> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
> budi->62815145150 in context default
> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/62815145150 at voiprakyat.or.id[1e55a5c7-5d58-4437-a48b-4278154b57a0]
> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/
> 62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA] [CALL_REJECTED]
>
>
> TIA
>
> budi
>
> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> In the mean time, I think it's a good article for you:
>>
>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>
>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>> > default configuration files of mod_sofia is in conf/sip_profiles
>> >
>> > http://wiki.freeswitch.org/wiki/Sofia
>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> >
>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>> >> thx, after reading the document i still got confusion about sofia
>> module
>> >> i give example below
>> >>
>> >> Dialing Through A Gateway(SIP Provider)
>> >>
>> >> A gateway is a means for making outbound calls through a SIP provider.
>> For
>> >> example:
>> >>
>> >> sofia/gateway/mygateway.com/1234
>> >>
>> >> by default i dont find any directory named sofia, where i should put
>> this
>> >> directory?
>> >> many document i read telling about sofia
>> >> TIA
>> >> budi
>> >>
>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
>> wrote:
>> >>>
>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>> >>>
>> >>> Yes I have siphit installed, I tried to make some changes on dialplan
>> file
>> >>> but call always goes to other server, what should I configure to
>> implement
>> >>> this outgoing call
>> >>>
>> >>>
>> >>>
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>> >
>> >
>> >
>> > --
>> > Blog: http://www.dujinfang.com
>> > Proj:  http://www.freeswitch.org.cn
>> >
>>
>>
>>
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