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-r...@voipconsulting.nl, Copyright (C) 1999 - 2008
Digium, Inc. and others.
Created by Mark Spencer <marks...@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-r...@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>
Call-ID: f264209-bccc3039@127.0.0.1
CSeq: 38885 REGISTER
Max-Forwards: 70
Contact: spaphone <sip:spaphone@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>
Call-ID: f264209-bccc3039@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@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>;tag=as6140f831
Call-ID: f264209-bccc3039@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@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>
Call-ID: f264209-bccc3039@127.0.0.1
CSeq: 38885 REGISTER
Max-Forwards: 70
Contact: spaphone <sip:spaphone@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>
Call-ID: f264209-bccc3039@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@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>;tag=as6140f831
Call-ID: f264209-bccc3039@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@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>
Call-ID: f264209-bccc3039@127.0.0.1
CSeq: 38885 REGISTER
Max-Forwards: 70
Contact: spaphone <sip:spaphone@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>
Call-ID: f264209-bccc3039@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@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@192.168.0.1>;tag=c2f457c6347c4f23o1
To: spaphone <sip:spaphone@192.168.0.1>;tag=as6140f831
Call-ID: f264209-bccc3039@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@127.0.0.1' in 32000 ms
(Method: REGISTER)
asterisk2*CLI> sip no debug
SIP Debugging Disabled
asterisk2*CLI>
--
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