asterisk users - Jul 2008 - what is the magic needed from upgrading from 1.4 to 1.6

I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working.

I am getting a SIP/401 Unauthorized error and then a SIP/404 error.
I changed nothing in the configs.

Is there a particular parameter needed for 1.6 that 1.4 did not care about?
If I drop back to 1.4 it starts working again.

Thanks

Jerry

Jerry Geis wrote:
> I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working.
> 
> I am getting a SIP/401 Unauthorized error and then a SIP/404 error.
> I changed nothing in the configs.
How are you getting SIP-related errors from Console/DSP? Posting a
console log would be most helpful, as many people on the mailing list
are not telepathic :-)

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

> ow are you getting SIP-related errors from Console/DSP? Posting a
> console log would be most helpful, as many people on the mailing list
> are not telepathic :-)
>
> -- 
> Kevin P. Fleming
> Director of Software Technologies
> Digium, Inc. - "The Genuine Asterisk Experience" (TM)
Kevin,
below is the log your talking about.

please note no configuration files were changed from 1.4  to 1.6, going back to
1.4 works again.

Jerry

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


Asterisk 1.6.0-beta9, 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':   == Found
[0;37;40m[1;30;40m  == [0;37;40mParsing '/etc/asterisk/extconfig.conf':
[1;30;40m  == [0;37;40mFound
Connected to Asterisk 1.6.0-beta9 currently running on ebox4300 (pid = 4877)
ebox4300*CLI> 
Verbosity is at least 5

[Kebox4300*CLI> 

<--- SIP read from UDP://192.168.1.8:5060 --->
INVITE sip:mediaport_audio_visual at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport
From: "Jerry Geis 204" <sip:3175661677 at
192.168.1.8>;tag=as7d1f7b71
To: <sip:mediaport_audio_visual at 192.168.1.25>
Contact: <sip:3175661677 at 192.168.1.8>
Call-ID: 1840b6730797640c3b4947535e878b62 at 192.168.1.8
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Jul 2008 16:53:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 20475 20475 IN IP4 192.168.1.8
s=session
c=IN IP4 192.168.1.8
t=0 0
m=audio 14322 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
?--- (14 headers 14 lines) ---
?  == Using SIP RTP CoS mark 5
?  == Using SIP VRTP CoS mark 6
?Sending to 192.168.1.8 : 5060 (NAT)
?Using INVITE request as basis request - 1840b6730797640c3b4947535e878b62 at
192.168.1.8
?No user '3175661677' in SIP users list
?Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from
192.168.1.8:5060
?
<--- Reliably Transmitting (no NAT) to 192.168.1.8:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.8:5060;branch=z9hG4bK029ea409;received=192.168.1.8;rport=5060
From: "Jerry Geis 204" <sip:3175661677 at
192.168.1.8>;tag=as7d1f7b71
To: <sip:mediaport_audio_visual at 192.168.1.25>;tag=as324df4b6
Call-ID: 1840b6730797640c3b4947535e878b62 at 192.168.1.8
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="0e961d2a"
Content-Length: 0


<------------>
?Scheduling destruction of SIP dialog '1840b6730797640c3b4947535e878b62 at
192.168.1.8' in 32000 ms (Method: INVITE)
?
[Kebox4300*CLI> 

<--- SIP read from UDP://192.168.1.8:5060 --->
ACK sip:mediaport_audio_visual at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport
From: "Jerry Geis 204" <sip:3175661677 at
192.168.1.8>;tag=as7d1f7b71
To: <sip:mediaport_audio_visual at 192.168.1.25>;tag=as324df4b6
Contact: <sip:3175661677 at 192.168.1.8>
Call-ID: 1840b6730797640c3b4947535e878b62 at 192.168.1.8
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (10 headers 0 lines) ---
?
<--- SIP read from UDP://192.168.1.8:5060 --->
INVITE sip:mediaport_audio_visual at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;rport
From: "Jerry Geis 204" <sip:3175661677 at
192.168.1.8>;tag=as7d1f7b71
To: <sip:mediaport_audio_visual at 192.168.1.25>
Contact: <sip:3175661677 at 192.168.1.8>
Call-ID: 1840b6730797640c3b4947535e878b62 at 192.168.1.8
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="devcentos5x64_to_ebox4300",
realm="asterisk", algorithm=MD5, uri="sip:mediaport_audio_visual
at 192.168.1.25", nonce="0e961d2a",
response="1a8e257ae008af4156b1f65be8d4d267"
Date: Mon, 21 Jul 2008 16:53:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 20475 20476 IN IP4 192.168.1.8
s=session
c=IN IP4 192.168.1.8
t=0 0
m=audio 14322 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
?--- (15 headers 14 lines) ---
?Sending to 192.168.1.8 : 5060 (NAT)
?Using INVITE request as basis request - 1840b6730797640c3b4947535e878b62 at
192.168.1.8
?No user '3175661677' in SIP users list
?Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from
192.168.1.8:5060
?Found RTP audio format 0
?Found RTP audio format 8
?Found RTP audio format 3
?Found RTP audio format 101
?Peer audio RTP is at port 192.168.1.8:14322
?Found audio description format PCMU for ID 0
?Found audio description format PCMA for ID 8
?Found audio description format GSM for ID 3
?Found audio description format telephone-event for ID 101
?Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
?Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
?Peer audio RTP is at port 192.168.1.8:14322
?Looking for mediaport_audio_visual in smvoice-mediaport (domain 192.168.1.25)
?
<--- Reliably Transmitting (no NAT) to 192.168.1.8:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.8:5060;branch=z9hG4bK6a460d62;received=192.168.1.8;rport=5060
From: "Jerry Geis 204" <sip:3175661677 at
192.168.1.8>;tag=as7d1f7b71
To: <sip:mediaport_audio_visual at 192.168.1.25>;tag=as324df4b6
Call-ID: 1840b6730797640c3b4947535e878b62 at 192.168.1.8
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
?[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: Call
from 'devcentos5x64_to_ebox4300' to extension
'mediaport_audio_visual' rejected because extension not found.
?Scheduling destruction of SIP dialog '1840b6730797640c3b4947535e878b62 at
192.168.1.8' in 32000 ms (Method: INVITE)
?
<--- SIP read from UDP://192.168.1.8:5060 --->
ACK sip:mediaport_audio_visual at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;rport
From: "Jerry Geis 204" <sip:3175661677 at
192.168.1.8>;tag=as7d1f7b71
To: <sip:mediaport_audio_visual at 192.168.1.25>;tag=as324df4b6
Contact: <sip:3175661677 at 192.168.1.8>
Call-ID: 1840b6730797640c3b4947535e878b62 at 192.168.1.8
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


<------------->
?--- (10 headers 0 lines) ---
?
[Kebox4300*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).

I dont see any errors in the dialplan while loading.
I tried to past the whole log but it was rejected.

Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files.

I cant even "dialplan show default" at this time.

Jerry