Best
Regards,
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-------Original
Message-------
Date:
22/03/2012 07:57:15
Subject:
Re: [Freeswitch-users] Gateway Configuration
(asteriskexampleprovided)
Fernando, here's the result:
send 662 bytes to udp/[189.84.133.130]:5060 at
10:49:40.725611:
------------------------------------------------------------------------
REGISTER sip:PROVIDER_IP;transport=udp
SIP/2.0
Via: SIP/2.0/UDP
SERVER_IP:5080;rport;branch=z9hG4bKFHjr44Ug1NZ7e
Max-Forwards: 70
From:
<sip:USERNAME@PROVIDER_IP;transport=udp>;tag=FN5Fjc2t9Be1K
To:
<sip:USERNAME@PROVIDER_IP;transport=udp>
Call-ID: ae19b5d8-740c-11e1-a151-17725ca17356
CSeq: 25870616 REGISTER
Expires: 3600
User-Agent:
FreeSWITCH-mod_sofia/1.0.head-git-914f6cb 2012-03-16
12-33-42 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
Supported: timer, precondition, path,
replaces
Content-Length: 0
------------------------------------------------------------------------
I've replaced the IPs according this aliases:
PROVIDER_IP -> My Provider IP
SERVER_IP -> My server IP (in which fs is
running)
Thank you very much for your help.
Anderson Arboleya
On Thu, Mar 22, 2012 at 7:45 AM, Fernando -
NextBilling IP Solutions
>
> Its Seem that FS cannot connect to your
gateway.
>
> Try the command "sofia profile external
siptrace on" and see the response from your gateway.
>
> Post here the siptrace, maybe i can help.
>
> Best regards,
> Importante:
> Esta mensagem, incluindo todo seu conteúdo,
contém informações confidenciais, legalmente
protegidas e destinadas a indivíduo e propósito
específicos. Caso a tenha recebido por engano,
lembramos do caráter sigiloso e solicitamos a
gentileza de desconsiderá-la e comunicar-nos o mais
breve possível.
> As informações contidas nesta mensagem e em
seu conteúdo são de responsabilidade de seu autor, não
representando necessariamente idéias, opiniões,
pensamentos ou qualquer forma de posicionamento por
parte da NextBilling IP Solutions.
> P "Antes de imprimir pense em seu compromisso
com o Meio Ambiente."
> -------Original Message-------
>
> From: Anderson Arboleya
> Date: 22/03/2012 07:40:53
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Gateway
Configuration (asterisk exampleprovided)
>
> Hi Fernando, I added the register param too.
>
> Brian, I've change the name to the SERVER_IP
my provider sent me.
>
> That's my configs now:
>
>
=======================================================
> #
/usr/local/freeswitch/conf/sip_profiles/external/testing.xml
> <include>
> <gateway name="SERVER_IP">
> <param name="username"
value="USERNAME" />
> <param name="realm"
value="SERVER_IP" />
> <param name="password"
value="PASSWORD" />
> <param name="register"
value="true"/>
> </gateway>
> </include>
>
> #
/usr/local/freeswitch/conf/dialplan/public/testing.xml
> <include>
> <extension name="testing-inbound">
> <condition field="destination_number"
expression="^(USERNAME)$">
> <action application="lua"
data="/root/call_logger.lua" />
> </condition>
> </extension>
> </include>
>
=======================================================
>
> And now I got this error:
>
>
=======================================================
> 2012-03-22 10:28:00.125504 [NOTICE]
sofia_reg.c:415 Registering 189.84.133.130
> 2012-03-22 10:28:32.125503 [ERR]
sofia_reg.c:1962 189.84.133.130
> Registration Failed with status Request
Timeout [408]. failure #1
>
=======================================================
>
> When I run a "sofia status" and the fs_cli I
got:
>
> external::SERVER_IP gateway
sip:USERNAME@SERVER_IP FAIL_WAIT
>
> Any other suggestions?
>
> Tks
>
> Anderson Arboleya
>
>
> > Whoops...
> >
> > By default it will try and register to
the gateway name unless stated
> > otherwise. So as your dialplan sits, its
registering to "testing" literally.
> > Should be "example.com" or the domain
you're registering to.
> >
> > -BDF
> >
> > On Mar 21, 2012 10:27 PM, "Anderson
Arboleya" <
me@nybras.com>
wrote:
> >>
> >> Hello,
> >>
> >> I'm trying to connect freeswitch to a
provider so I can call real
> >> numbers and reach my freeswitch
server, but I'm kinda stuck in this
> >> process.
> >>
> >> My provider sent me this Asterisk's
configs:
> >>
> >>
=======================================================
> >> # sip.conf:
> >> register =>
USERNAME:PASSWD@SERVER_IP
> >>
> >> [directcall12]
> >> context=default
> >> fromuser= USERNAME
> >> username= USERNAME
> >> secret= PASSWD
> >> type=friend
> >> host=SERVER_IP
> >> canreinvite=no
> >> nat=no
> >> dtmfmode=rfc2833
> >> insecure=very
> >> disallow=all
> >> allow=g729
> >>
> >> # extensions.conf:
> >> exten => 3131,1,Noop(ORIGEM -
${CALLERID(num)})
> >> exten =>
3131,n,Dial(${TRUNKPBX}/${RAMAL})
> >> exten => 3131,n,Hangup()
> >>
> >> ** change these settings according
you internal extensions configs
> >>
=======================================================
> >>
> >>
> >> Well, I tried adapting it to
FreeSwitch like this (according some
> >> examples I found across the web):
> >>
> >>
=======================================================
> >> #
/usr/local/freeswitch/conf/sip_profiles/external/testing.xml
> >> <include>
> >> <gateway name="testing">
> >> <param
name="username" value="USERNAME" />
> >> <param name="realm"
value="SERVER_IP" />
> >> <param
name="password" value="PASSWORD" />
> >> </gateway>
> >> </include>
> >>
> >> #
/usr/local/freeswitch/conf/dialplan/public/testing.xml
> >> <include>
> >> <extension
name="testing-inbound">
> >> <condition
field="destination_number"
expression="^(USERNAME)$">
> >> <action application="lua"
data="/root/call_logger.lua" />
> >> </condition>
> >> </extension>
> >> </include>
> >>
=======================================================
> >>
> >> In the end all I got is this log:
> >>
> >> 2012-03-17 01:41:23.565471 [NOTICE]
sofia_reg.c:415 Registering testing
> >> 2012-03-17 01:41:55.565460 [ERR]
sofia_reg.c:1962 testing Registration
> >> Failed with status Request Timeout
[408].
> >>
> >> All I want is to connect with my
provider and route the calls made to
> >> the real number my provider gave me
to my Dialplan, so my lua script
> >> will be called.
> >>
> >> Btw, I dont actually bought a
"dedicated number" but an extension. I
> >> mean, I will call a number and an
IVR-menu will ask me for an
> >> extension, then I type the extension
I bought (3131) and their system
> >> will send the call for me. Is this
clear?
> >>
> >> Sorry, I'm an experienced programmer
but I'm just starting with all
> >> this telephony thing and I don't know
what else I need to do from this
> >> point on, I have searched a lot and
tried a lot of different
> >> approaches, but none succeed.
> >>
> >> Could anybody help me with this,
please?
> >>
> >> Thanks in advance.
> >>
> >>
> >> Anderson Arboleya
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