On 7/2/2010 8:55 AM, Michael Scheidell wrote:
Only direct mention in google searches for sipx+PAP2 show a wiki that
says it should work, and people asking for sipx support in the GUI.
is it close enough to one of the other Linksys TA's so that we can use
it? or do I need to create my own xml files and load them (where? in
tftp?)
and, what firmware to work best with sipx 4.2.0.
will NOT be using these remote. just need to move them from old
broadsoft hosted platform where they controlled a couple of security
door/cameras over to sipx.
I seem to have a PAP2T with fw 3.19(Lsc) and a PAP2 with fw 2.0.12(LS)
It is easy enough to log into the web interface for the unit and
configure the account information. You really don't need to change
anything else. Faxing from the unit works great with voip.ms. I
couldn't get faxing to work with ht286. The default firmware worked
fine, but I decided to upgrade to the latest (5.1.6) which works good
also. You have to use a windows program that comes with the firmware
to upload the firmware to the unit. Here are the release notes on 5.1.6:
========================================
New Features
========================================
--- External Notes ---
- If REGISTER results in a 301 response with a Contact header
that has a maddr URI parameter, and if the <Outbound Proxy>
is an IP address, the SPA will change the outproxy proxy
address to the value of the maddr address. This value will
remain valid until the next 301 response, if any, or will
restore to the originally configured value upon reboot.
- When swapping calls in a call-waiting or similar situation,
the SPA will order the operations to make sure that call hold
is invoked before call resume.
- New RC blink pattern. Power LED red-orange slow blink while
contacting server. Red-orange double blink if file not found
or corrupt.
- 1. Make DHCP DISCOVER retry longer (120 Seconds)2. Reboot the
device in 10 minutes if the device is using last konwn good
IP address.
========================================
Bug Fixes
========================================
--- External Notes ---
- SDP in INVITE response, includes annexb=yes/no if present in
INVITE message
- Fixed this problem: A,B,C,D digits are dropped when dialing
- SPA should escape occurrences of '#' in the dial string with
%23 in outbound INVITE
- Fixed this problem: Unit stops playing incoming RTP audio once
it receives a 180/183 w/o SDP; regardless setting of <Sticky 183>
parameter
- bug fix: Bogus IP may be used if dns srv failed when querying
the proxy server.
- Worked around this problem: some server will change the Contact
header's address in their response to the SPA's SIP REGISTER
request, so that the SPA cannot find the corresponding Contact
in the response and therefore may not be able to extract the
proper expires value inserted by the server. This is a problem
if the expires value inserted by the server in the response is
smaller than the value in the original SIP REGISTER request.
The work around is to use the first entry in the Contact header
if an exactly matching address is not found.
- Fixed this problem: If unknown response is received, such as 101,
SPA might misinterpret it as a 6xx response. In some other cases,
an unknown response, such as 188, might be dropped by the SPA as
a bad message. The proper way is to treat it as the corresponding
base response, such as 100.
- Fixed this problem: Daylight Saving Time is not correct if start
time is later then end time of the year in the <Daylight Saving
Time Rule> parameter
- Fixed this problem: Unit should ignore in-dialog re-INVITE if it
has not received ACK to the 200 response to the initial INVITE
from the caller. This condition might happen if the ACK was lost
but the first re-INVITE has already been sent by the peer.
- Fixed this problem: Unit should not filter out leading * or #
digit in the incoming caller id number, as these can be legitimate
phone number digits
- Fixed this problem: PAP2T cannot detect DTMF digits at 50ms on/off
during a call, even if <DTMF Tx Mode> is "normal".
- Fixed missing '*' and '#' in on-hook caller-id number generation
- Eliminated cause of spurious reboot on resync for customized units
with read-only parameters and value mismatch.
--
Regards
--------------------------------------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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