asterisk users - Jul 2011 - Registration problems, Linksys SPA 3102 on Asterisk 1.4.20

Sorry, I am resending this, I tried earlier, but I 
couldn't see it appear on the archives - 
apologogies if it appears double!

--------------------------------------------------

My Sipura 3000 ATA died on me this morning. I had 
a Linksys SPA 3102 available which I would like to 
use as a replacement. Unfortunately, the SPA3102 
is not able to register with the asterisk server - 
I am always getting a
> SIP/2.0 401 Unauthorized

response from the asterisk server upon which the 
SPA3102 unit reports on its web interface that the 
registration FAILED.

* I checked (and double checked) whether the given 
credentials stored in the SPA unit match the 
required ones as defined in the server's sip.conf 
file, and they do match.

* I also upgraded the SPA's firmware from the 
older 3.6 version to 5.1.10 (GW)

* During my research in different forums I found 
some posts that hinted on using TCP instead of UDP 
for SIP transport, so I set
> PSTN Line -> SIP Settings -> SIP Transport
to
> TCP
but no success.

I still get the 401 error response from the 
server. I wonder how to get the SPA unit to 
successfully register with my asterisk server.

1. Are there any settings (either on the SPA unit 
or the asterisk server) which I have overlooked?

2. Is there some sort of compatibility issue 
between the SPA3102 and asterisk 1.4.20?

Below I posted some more details. Thank you so 
much for your consideration, help is very much 
appreciated!

Peter Hoppe



1. sip.conf
============
> [general]
> ;
---------------------------------------------------------------------------------
> ; 1.1    General setup
> ;
> bindaddr        = 192.168.0.1
> port            = 5060
> tos             = none
>
> ;
---------------------------------------------------------------------------------
> ; 1.2    Jitter buffer configuration
> ;
>
> ;
---------------------------------------------------------------------------------
> ; 1.3    Codecs setup
> ;
> disallow        = all
> allow           = alaw
>
> ;
---------------------------------------------------------------------------------
> ; 1.4    Other options
> ;
> context         = default
> defaultexpirey  = 160
> dtmfmode        = info
> maxexpirey      = 180
> nat             = never
> qualify         = no
> record-in       = On-Demand
> record-out      = On-Demand
> type            = friend
>
> ;
---------------------------------------------------------------------------------
> ; 2      Devices for their respective contexts
> ;
> [spaphone]
> accountcode     = spaphone
> callerid        = spaphone
> canreinvite     = yes
> context         = pstn
> dtmfmode        = info
> host            = dynamic
> mailbox         > port            = 5060
> qualify         = yes
> secret          = abcde
> type            = friend
> username        = spaphone



2. Asterisk version:
============
> Asterisk 1.4.20-1 RPM by vc-rpms at voipconsulting.nl, Copyright (C) 1999 -
2008 Digium, Inc. and others.
> Created by Mark Spencer <markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show
warranty' for details.
> This is free software, with components licensed under the GNU General
Public
> License version 2 and other licenses; you are welcome to redistribute it
under
> certain conditions. Type 'core show license' for details.
>
========================================================================>  
== Parsing '/etc/asterisk/asterisk.conf': Parsing
/etc/asterisk/asterisk.conf
> Found
> Connected to Asterisk 1.4.20-1 RPM by vc-rpms at voipconsulting.nl
currently running on asterisk2 (pid = 2336)
> Verbosity is at least 5
> Core debug is at least 1


3. spa-3102 details:
============
> Product Name: SPA-3102
> Software Version: 5.1.10(GW)
> Hardware Version: 1.4.5(a)
> LAN IP address: 192.168.0.10
> LAN subnet mask: 255.255.255.0
>
> PSTN Line -> SIP settings
>     SIP Transport: UDP
>     SIP Port: 5060
> PSTN Line -> Proxy and Registration
>     Proxy: 192.168.0.1
> PSTN Line -> Subscriber information
>     Display name: spaphone
>     User ID: spaphone
>     Password: abcde



4. SIP debug output on asterisk console:
============> REGISTER sip:192.168.0.1 SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> Max-Forwards: 70
> Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;expires=3600
> User-Agent: Linksys/SPA3102-5.1.10(GW)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura, replaces
>
>
> <------------->
> --- (12 headers 0 lines) ---
> Using latest REGISTER request as basis request
> Sending to 127.0.0.1 : 5060 (no NAT)
> asterisk2*CLI>
> <--- Transmitting (no NAT) to 127.0.0.1:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:spaphone at 192.168.0.1>
> Content-Length: 0
>
>
> <------------>
> asterisk2*CLI>
> <--- Transmitting (no NAT) to 127.0.0.1:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>;tag=as6140f831
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1aa1d724"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'f264209-bccc3039 at
127.0.0.1' in 32000 ms (Method: REGISTER)
> asterisk2*CLI>
> <--- SIP read from 192.168.0.10:5060 --->
> REGISTER sip:192.168.0.1 SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> Max-Forwards: 70
> Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;expires=3600
> User-Agent: Linksys/SPA3102-5.1.10(GW)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura, replaces
>
>
> <------------->
> --- (12 headers 0 lines) ---
> Using latest REGISTER request as basis request
> Sending to 127.0.0.1 : 5060 (no NAT)
> asterisk2*CLI>
> <--- Transmitting (no NAT) to 127.0.0.1:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:spaphone at 192.168.0.1>
> Content-Length: 0
>
>
> <------------>
> asterisk2*CLI>
> <--- Transmitting (no NAT) to 127.0.0.1:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>;tag=as6140f831
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1aa1d724"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'f264209-bccc3039 at
127.0.0.1' in 32000 ms (Method: REGISTER)
> asterisk2*CLI> exit
> <--- SIP read from 192.168.0.10:5060 --->
> REGISTER sip:192.168.0.1 SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> Max-Forwards: 70
> Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;expires=3600
> User-Agent: Linksys/SPA3102-5.1.10(GW)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura, replaces
>
>
> <------------->
> --- (12 headers 0 lines) ---
> Using latest REGISTER request as basis request
> Sending to 127.0.0.1 : 5060 (no NAT)
> asterisk2*CLI> exit
> <--- Transmitting (no NAT) to 127.0.0.1:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:spaphone at 192.168.0.1>
> Content-Length: 0
>
>
> <------------>
> asterisk2*CLI> exit
> <--- Transmitting (no NAT) to 127.0.0.1:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10
> From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1
> To: spaphone <sip:spaphone at 192.168.0.1>;tag=as6140f831
> Call-ID: f264209-bccc3039 at 127.0.0.1
> CSeq: 38885 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1aa1d724"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'f264209-bccc3039 at
127.0.0.1' in 32000 ms (Method: REGISTER)
> asterisk2*CLI> sip no debug
> SIP Debugging Disabled
> asterisk2*CLI>