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 Contact: <sip:gw+PROVIDER_IP@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@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.
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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
On Wed, Mar 21, 2012 at 11:41 PM, Brian Foster <bdfoster@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@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