401 Unauthorized to all phones

General Help
1

Hi there!

I am setting up a new FreePBX system to replace another that I am having lots of problems with.

I used the CentOS image downloaded from the site the other day.

I have setup two test extensions and a few test conferences. Neither of my extensions can connect and register to the server. They can ping the server, and I see the rejection on both sides as a 401 unauthorized.

I have tried from a Snom 370 (latest version 8 firmware) and from a 3CX softphone on my PC.

I have confirmed that the secret is correct for both extensions. Although the Snom phone does ask for the SIP password.

The extensions in the PBX are CHAN_SIP Driver extensions, I have also tried CHAN_PJ SIP Driver. Neither has proved to work.

I have also matched the settings to my current PBX for all of the extension options. They are left pretty much to the default.

I recognize that this is a beta level build, but I have no functionality at this moment.

Thanks!

Version Info:
PBX Firmware: 6.12.65-9
PBX Service Pack: 1.0.0.0

CentOS version: 6.5 (Final)
Asterisk Version: 12.1.1

Modules:
Admin

Custom Applications 2.11.0.0 Stable FreePBX GPLv2+ Enabled
Digium Addons 2.8.1 Stable Digium GPLv2 Enabled
Feature Code Admin 2.11.0.0 Stable FreePBX GPLv2+ Enabled
FreePBX ARI Framework 2.11.0.4 Stable FreePBX GPLv2+ Enabled
FreePBX Framework 12.0.1alpha40 Stable FreePBX GPLv2+ Enabled
System Admin 2.11.0.50 Stable Schmoozecom.com Commercial Enabled
iSymphony 1.4.8 Stable Enabled
Applications

Call Recording 2.11.0.4 Stable FreePBX GPLv2+ Enabled
Core 12.0.1alpha6 Stable FreePBX GPLv2+ Enabled
Info Services 2.11.0.1 Stable FreePBX GPLv2+ Enabled
Connectivity

Digium Phones Config 2.10.0.9 Stable Digium GPLv2 Enabled
Internal Options & Configuration

Conferences 2.9.0.2 Stable FreePBX GPLv2+ Enabled
Recordings 3.3.11.2 Stable FreePBX GPLv2+ Enabled
Reports

Asterisk Logfiles 2.11.0.7 Stable Schmooze Com. Inc. GPLv2+ Enabled
CDR Reports 2.11.0.4 Stable FreePBX GPLv2+ Enabled
System Dashboard 2.12.0.1 Stable FreePBX GPLv2+ Enabled
Settings

Asterisk SIP Settings 12.0.1alpha2 Stable schmoozecom.com AGPLv3 Enabled
Music on Hold 2.11.0.1 Stable FreePBX GPLv2+ Enabled
Voicemail 2.11.1.1 Stable FreePBX GPLv2+ Enabled

2

Phone Side:

6/4/2014 16:21:00 [NOTICE] PHN: SIP: process auth:Match challenge for user=1036, realm=asterisk
16/4/2014 16:21:13 [WARN ] PHN: SIP: process_registrar_packet: 401 needs 128 bit nonce

Server Side:

[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:96) - Undefined offset: 1
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:97) - Undefined offset: 2
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:98) - Undefined offset: 3
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:99) - Undefined offset: 4
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:100) - Undefined offset: 5
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:96) - Undefined offset: 1
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:97) - Undefined offset: 2
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:98) - Undefined offset: 3
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:99) - Undefined offset: 4
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:100) - Undefined offset: 5
[2014-Apr-16 16:00:02] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/storage.php:35) - Undefined index: sysadmin_exp
[2014-Apr-16 16:19:01] [PHP-WARNING] (/var/www/html/admin/modules/digiumaddoninstaller/functions.inc.php:23) - include(modules/digiumaddoninstaller/libdregister/digium_register.php): failed to open stream: No such file or directory
[2014-Apr-16 16:19:01] [PHP-WARNING] (/var/www/html/admin/modules/digiumaddoninstaller/functions.inc.php:23) - include(modules/digiumaddoninstaller/libdregister/digium_register.php): failed to open stream: No such file or directory
[2014-Apr-16 16:19:01] [PHP-WARNING] (/var/www/html/admin/modules/digiumaddoninstaller/functions.inc.php:23) - include(): Failed opening ‘modules/digiumaddoninstaller/libdregister/digium_register.php’ for inclusion (include_path=’.:/usr/share/pear:/usr/share/php’)
[2014-Apr-16 16:23:33] [PHP-NOTICE] (/var/www/html/admin/libraries/view.functions.php:6) - Undefined variable: db
[2014-Apr-16 16:23:34] [PHP-NOTICE] (/var/www/html/admin/modules/logfiles/views/logs.php:7) - Undefined variable: full
[2014-Apr-16 16:23:35] [PHP-NOTICE] (/var/www/html/admin/libraries/view.functions.php:6) - Undefined variable: db
[2014-Apr-16 16:23:39] [PHP-NOTICE] (/var/www/html/admin/libraries/view.functions.php:6) - Undefined variable: db

3

Are these phones on the same subnet as the PBX? Can you post your Asterisk SIP Settings? Mainly the top part with localnet and externip settings.

4

Phones are on the same subnet as the server. I run a /22 network, so 192.168.100.0 - 192.168.103.255. Static IPs are in the .100 network, DHCP for the .101 - .103

The server is in the .100 network. My hard phone (Snom 370) is in DHCP range, and has a .101 IP. My desktop is static, and has an IP in the .100 range. So my softphone is in the desktop.

Grabs of sip settings found here:
http://imgur.com/u5Yl4PU
http://imgur.com/ZdeyGJ6

Thanks!

5

You got the bonus plan, the entire CODEC section.

Should not be any NAT issues on that huge 767 host network.

6

What NAT setting do you have your extensions set to?

7

Finally! The ability to post again!

Phone NAT settings here:
http://imgur.com/OGXFEZF

Problem server NAT settings (the broken one):
http://goo.gl/ZtQU3S

Current Good server (same subnet as new one):
http://goo.gl/DllB70

Let me know if there is anything else I pull for you!

8

I have found that I cannot respond or update unless I use Internet Explorer.

I use IE9 right now for this, just in case anyone actually wants to help out on this subject.

Otherwise it is Chrome all the way.

9

Is more information needed?

Does this need to be bumped?

Can anyone help me?

10

Did you solve this in the end? If not it might be worth posting the relevant parts of your Asterisk log file. So far you’ve only posted the FreePBX log file.