[Freeswitch-users] outgoing gateway
David Ponzone
david.ponzone at gmail.com
Sun May 9 02:49:08 PDT 2010
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
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
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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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