[Freeswitch-users] Gateway Configuration (asterisk exampleprovided)

Thu Mar 22 13:55:43 MSK 2012

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 at PROVIDER_IP;transport=udp>;tag=FN5Fjc2t9Be1K
   To: <sip:USERNAME at PROVIDER_IP;transport=udp>
   Call-ID: ae19b5d8-740c-11e1-a151-17725ca17356
   CSeq: 25870616 REGISTER
   Contact: <sip:gw+PROVIDER_IP at SERVER_IP:5080;transport=udp;gw=189.84.133.130>
   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
<fernandojdk at gmail.com> wrote:
>
> 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:
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> -------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 at SERVER_IP   FAIL_WAIT
>
> Any other suggestions?
>
> Tks
>
> Anderson Arboleya
>
>
> On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> > 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 at 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 at 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