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[pabx_freeswitch at tel...]

I want to register a telephone 2010 (a remote extension) to my FS server.

My server has dynamic ip address, registered with DynDNS. (dyndns ip
address is: xxxx.ftpaccess.cc)
In the remote telephone I programmed:
a. stun server=stun.freeswitch.org
b. sip port=5090 (i replaced all the 5060 with 5090 at several places)
c. rtp port=17000
d. domain= xxxx.ftpaccess.cc (my dyndns)

The remote phone is also on a dynamic ip address.


I followed the instructions on
http://wiki.freeswitch.org/wiki/Example_Offsite_phones

Step 1. I created an extension 2010 in /..../directory/default.xml
=================================================
<include>
user id="2010" mailbox="2010">
params>
param name="password" value="2010"/>
param name="vm-password" value="2010"/>
/params>
variables>
variable name="accountcode" value="2010"/>
variable name="user_context" value="default"/>
variable name="effective_caller_id_name" value="Terje"/>
variable name="effective_caller_id_number" value="2010"/>
/variables>
/user>
</include>


Setp 2: I created a new profile. called "doublenat" profile set to port
5090 in /..../sip_profiles/doublenat.xml
==================================================================================
<profile name="doublenat">
gateways>
X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
/gateways>
settings>
param name="debug" value="0"/>
param name="sip-trace" value="no"/>
param name="rfc2833-pt" value="101"/>
param name="sip-port" value="5090"/>
param name="dialplan" value="XML"/>
param name="context" value="public"/>
param name="dtmf-duration" value="100"/>
param name="codec-prefs" value="$${outbound_codec_prefs}"/>
param name="use-rtp-timer" value="true"/>
param name="hold-music" value="$${hold_music}"/>
param name="rtp-timer-name" value="soft"/>
param name="manage-presence" value="false"/>
param name="aggressive-nat-detection" value="true"/>
param name="apply-nat-acl" value="rfc1918"/>
param name="inbound-codec-negotiation" value="generous"/>
param name="nonce-ttl" value="60"/>
param name="auth-calls" value="false"/>
param name="rtp-timeout-sec" value="1800"/>
param name="rtp-ip" value="$${local_ip_v4}"/>
param name="sip-ip" value="$${local_ip_v4}"/>
param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
param name="ext-sip-ip" value="$${external_sip_ip}"/>
param name="rtp-timeout-sec" value="300"/>
param name="rtp-hold-timeout-sec" value="1800"/>
/settings>
</profile>


Step 3: Inside the /..../conf/dialplan/public.xml, I put
=========================================
<extension name="public_extensions">
<condition field="destination_number" expression="^(20[01][0-9])$">
<action application="bridge" data="sofia/doublenat/$1%$${domain} "/>
</condition>
</extension>


However the phone can't register.

In the FS CLI window I get:

2008-09-24 20:10:39 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't
find user [2010@xxxx.ftpaccess.cc]
You must define a domain called 'henkoegema.ftpaccess.cc' in your directory
and add a user with the id="2010" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
%-|

This line repeats and repeats and repeats.

Evenso the warning is clear, I'm not sure what exactly I should do. (and
where) :confused:

Henk
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[pabx_freeswitch at tel...]

I still haven't been able to solve my problem.

Still getting:
=========
2008-09-26 11:02:59 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't
find user [2010@xxxx.ftpaccess.cc]
You must define a domain called 'xxxx.ftpaccess.cc' in your directory and
add a user with the id="2010" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2008-09-26 11:03:37 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't
find user [2010@xxxx.ftpaccess.cc]
You must define a domain called 'xxxx.ftpaccess.cc' in your directory and
add a user with the id="2010" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
...............etc.....etc....etc

Have no idea how to solve this. :rules:

Henk




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[anthony.minessale at g...]

Doesn't the long winded error message even give you a hint?

I am quite suprised by how much difficulty people seem to have with this concept.

SIP and FS are both domain based. Just like email, IM and most other internet protocols.

If you are pointing a sip client at FS, you need to specify the same domain in your sip phone that you have FS
configured to be in charge of. It's like if you wanted to go to our website you must point your browser at
freeswitch.org or you will not get there. The IP that leads to freeswitch.org may lead to 200 websites and the
domain name is the only realy way to tell where you want to go.
If you just tell the sip client the address of FS and FS is set to manage a specific IP or domain name it will not be able to find the user.

like the error says, edit your directory and make sure the user and domain in your directory config and the user and domain in your sip client match, that's all that's to it.

On Fri, Sep 26, 2008 at 4:12 AM, henkoegema <pabx_freeswitch@telenet.be (pabx_freeswitch@telenet.be)> wrote:

[anthony.minessale at g...]

Anthony Minessale-2 wrote:

[anthony.minessale at g...]

close enough,
more importantly let it install all the defaults again.

mv /usr/local/freeswitch /usr/local/freeswitch.bak

and make install again to get the latest defaults


On Sat, Sep 27, 2008 at 4:23 PM, henkoegema <pabx_freeswitch@telenet.be (pabx_freeswitch@telenet.be)> wrote:

[anthony.minessale at g...]

yes

On Sat, Sep 27, 2008 at 5:01 PM, henkoegema <pabx_freeswitch@telenet.be (pabx_freeswitch@telenet.be)> wrote:

[anthony.minessale at g...]

assuming you are not checking out to the same dir you installed it too.
and assuming you are not repeating that checkout command over and existing install.

once you check out the trunk you should do subsequent upgrades with
"make current"


On Sat, Sep 27, 2008 at 5:01 PM, henkoegema <pabx_freeswitch@telenet.be (pabx_freeswitch@telenet.be)> wrote: