[Freeswitch-users] outgoing gateway

Sun May 9 15:42:24 PDT 2010

dear all
thx for your support it's working now :)
next question is after i change some xml file then i do reloadxml, but in
fact not all config changes.
reason of my statement is: many time i do reload xml but seems my config
still not work, then i shutdown the FS, then invoke ./freeswitch again and
work
any advice?

regards

budi

On Mon, May 10, 2010 at 5:06 AM, budi wibowo <bwibowo at gmail.com> wrote:

> 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
>> 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 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
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Blog: http://www.dujinfang.com
>>>> Proj:  http://www.freeswitch.org.cn
>>>>
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