[Freeswitch-users] outgoing gateway

budi wibowo bwibowo at gmail.com
Sun May 9 15:06:34 PDT 2010 

thx a lot,
refering to http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
with required changes but still failed, i also attached the F8 capture, hope
can solve this problem


regards
budi

<http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>

On Sun, May 9, 2010 at 4:49 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> You need to add this in your gateway config:
>
>   <param name="username" value="none"/>
>   <param name="password" value="none"/>
>
> and change the register param to:
>           <param name="register" value="false"/>
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
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>
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> d'altération. **IPeva** décline toute responsabilité au titre de ce
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> destinataire de ce message, merci de le détruire immédiatement et d'avertir
> l'expéditeur.*
> *
> *
>
>
>
> Le 09/05/2010 à 02:16, budi wibowo a écrit :
>
> 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
>>
>> 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 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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>>> >>
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>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Blog: http://www.dujinfang.com
>>> > Proj:  http://www.freeswitch.org.cn
>>> >
>>>
>>>
>>>
>>> --
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