SOLVED SIP/2.0 401 Unauthorized, when registering (older works!)

[ksDevGuy]

Ok I am stumped! Trying to upgrade from my Asterisk 1.8.13.0/FreePBX2.11 install to the Oracle Virtual version of the latest Asterisk 1.11.x/FreePBX2.11 ...

However -- all my Cisco 7960 & 7971's get the "SIP/2.0 401 Unauthorized" SIP response when trying to register! My SPA & Siemens SIP devices (not using t*f*t*p) register fine, but not the Cisco's (and I have manually entered settings into the 7960's - same result, no registration).

What is strange, is that if I pull the ether cable and swap the two servers -- my old server handles the phones perfectly (as it always has of course). The new host/virtual server won't register a single one (I matched passwords to shorter ones, have dropped IP tables, Fail2Ban, the host's Windows firewall -- all to no avail).

I'm at wits end -- why the new distro (albeit in a Oracle VM in this case) won't register when the previous v2.0.x distro does (baremetal) is killing me .... help?

Thanks,
ksDevGuy

 

[rossiv]

Are you using TCP or UDP transport for those?

 

[ksDevGuy]

Both installs have FreePBX defaulting to UDP, which I understand to mean it can use TCP when needed for a handshake, etc. but pursues UDP ultimately (I also could not understand this quite right as well!).

 

[ksDevGuy]

Any ideas from the peanut gallery? Killing me I can't make the jump to a slightly newer PIAF due to this simple issue!

Thanks gang,
ksDevGuy

 

[Hyksos]

http://pbxinaflash.com/community/in...7965-fails-to-register-with-piaf-green.13885/

 

[phonebuff]

What IP address (Asterisk Server) are the config files on the t*f*t*p server pointing at ?

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[Hyksos]

NOT saying you have exactly the same issue.

 

[ksDevGuy]

SOLUTION

Asterisk 1.11+ & Cisco 79xx handsets
====================================
- In FreePBX "Settings -> SIP Settings" screen and set NAT to "off" at the top of the screen
- Make sure each extension has NAT off
- Make sure extension password is 31 characters or less (i.e. FreePBX automatic guid passwords are 32 characters ... 1 character too long!)

That's it! Have fun all.

ksDevGuy

 

[asummerell]

ksDevGuy,

Just to confirm, you got those settings to work for the PBX on the same LAN, correct? What about an external (hosted) PBX? I can get our 7961 and 7962 models to register and work properly locally by turning off NAT.

However, unless I'm missing something, you need to have NAT enabled to have multiple phones register to an external (hosted) PBX? Otherwise the PBX will not be able to send requests to the individual phones.

I have only been able to get a single extension to register against a hosted PBX by turning off NAT, and then port forwarding UDP 5060 in the router where the phone is located to the phone's internal IP. When that happens though, peer details in asterisk show the phone's internal IP address for the host instead of the external IP of the phone.
*Update to that paragraph: I was able to get 2 way audio by port forwarding my RT P range to the internal IP address of the phone, and in the SEP<MAC>.cnf.xml file using the following:
<natEnabled>true</natEnabled>
<natAddress>EXTERNAL-IP-ADDRESS</natAddress>

That is all good and fine if we have access to be able to port forward, if there is only one phone on the network, and if there is a static external IP. However, what can we do for a setup where there is say 20 phones?

Any ideas?

Thanks!