so your extension are not registered

On my FS I can see my own extension.

Now this is where I am going to learn something because I have no idea why you can call from one extension to another if they are not registered.

On 13-05-22 10:48 PM, Mike Hendrie wrote:

Sorry it took so long. 

I updated the log file and your requested details. 

http://pastebin.freeswitch.org/20963

Thank you

On Wed, May 22, 2013 at 10:38 AM, Michael Collins <msc@freeswitch.org> wrote:

Mike,

You might be confusing "profiles" with "domains". A profile is a "SIP profile" and represent a SIP user agent where FreeSWITCH's SIP stack ("Sofia") listens and responds. There is no explicit relationship between domains and profiles. Profiles simply define where SIP messages come and go on the system.

Domains are a logical construct in the server. You can have multiple domains and thus multiple users with the same ID, i.e. "1001@domain1" and "1001@domain2".

Go to your fs_cli and capture the output of these commands:

sofia status

sofia status profile internal reg

Drop those in a pastebin a link back here.

-MC

On Wed, May 22, 2013 at 7:14 AM, Mike Hendrie <mike@hendrienet.com> wrote:

I will look into the registration of the phones as I see I am missing something.

freeswitch@internal> sofia status profile GothamCity.xom reg

Registrations:

=================================================================================================

Total items returned: 0

=================================================================================================

On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie <mike@hendrienet.com> wrote:

I do.  I can also ring extension 1000.

On May 22, 2013 7:52 AM, "Philippe Le Toquin" <philippe@ppmt.org> wrote:

This is why it goes to voicemail I guess

1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099 sofia/external/5555555555@66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[1001] to event
2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044 Parsing global variables
3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[GothamCity.xom]
4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[1001@GothamCity.xom]
5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106 Originate Failed.  Cause: USER_NOT_REGISTERED
10. EXECUTE sofia/external/5555555555@66.66.66.66 answer()
11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772 Set Codec sofia/external/5555555555@66.66.66.66 P

Do you have a dialtone on that 1001 extension?

On 22 May 2013 07:55, Mike Hendrie <mike@hendrienet.com> wrote:

Thank you.

Here is the log URL.

http://pastebin.freeswitch.org/20960

On Wed, May 22, 2013 at 1:39 AM, Michael Collins <msc@freeswitch.org> wrote:

Post a FreeSWITCH debug log of the incoming call. Use pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax highlighting. Paste the URL in this email thread and we'll take a look.

-MC

On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie <mike@hendrienet.com> wrote:

Correction:

I had a second dialplan in the public folder that was causing confusion. Below is the dialplan I am using.

If I change the extension in the dialplan from 1000 to 1001 I get the appropriate voice mail extension, however, the phones never ring.

I have the fs configured as a multi-tenant solution. 

Could the dialplan be using the default extensions (1000 and 1001) under /conf/directory/default and not reference the /conf/directory/GothamCity.xom domain? That would explain why I get to the voicemail for the correct extension when the phone never rings.

Thanks

=====

 /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml

<include>

  <extension name="GothamCity.xom 1001">

    <condition field="destination_number" expression="^1?(2624481175)$">

       <action application="set" data="domain_name=GothamCity.xom"/>

       <action application="transfer" data="1001 XML default"/>

    </condition>

  </extension>

</include>

=====

/conf/directory/GothamCity.xom.xml

  <domain name="GothamCity.xom">

    <params>

      <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>

    </params>

    <variables>

      <variable name="record_stereo" value="true"/>

      <variable name="user_context" value="default"/>

      <variable name="default_gateway" value="$${default_provider}"/>

      <variable name="default_areacode" value="$${default_areacode}"/>

      <variable name="transfer_fallback_extension" value="operator"/>

    </variables>

    <groups>

      <group name="GothamCity.xom">

        <users>

          <X-PRE-PROCESS cmd="include" data="GothamCity.xom/*.xml"/>

        </users>

      </group>

      <group name="sales">

        <users>

          <!--

              type="pointer" is a pointer so you can have the

              same user in multiple groups.  It basically means

              to keep searching for the user in the directory.

          -->

          <user id="1000" type="pointer"/>

          <user id="1001" type="pointer"/>

          <user id="1002" type="pointer"/>

          <user id="1003" type="pointer"/>

          <user id="1004" type="pointer"/>

        </users>

      </group>

=====

On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie <mike@hendrienet.com> wrote:

Thank you for your assistance. I made the suggested modification below, however, when calling the number it goes directly to voicemail.

  /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml

<include>

  <extension name="vitel-inbound">

    <condition field="destination_number" expression="^1?(262xxxxxxx)$">

       <action application="transfer" data="1000 XML default"/>

    </condition>

  </extension>

</include>

<snip>

--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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_________________________________________________________________________
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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