[Asterisk-Users] 480i User Feedback With Asterisk (fwd)

[jparr at bgcfreedom.com]

For those that are interested, here is my report back to Sayson on the
480i

---------- Forwarded message ----------
Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT)
From: xxx at bgcfreedom.com
To: xxx at sayson.com
Subject: 480i User Feedback With Asterisk

Seshu,

I am using a 480i, and I am impressed with the phone on a whole, but
obviously the firmware is lacking. Details follow.

Hold button works, but holds the user at the phone, does not hold them at
the PBX, allowing for music on hold. I would also like to see the hold
button software addressable so that it could be used to park the call
(transfer to the parking extension) rather than only putting the call on
hold.

Transfer seems to work fine, but would like to see blind transfer
(transfer direct to the remote party, rather than the current behavior
where you are bridged to the called party, then have to press transfer
again to complete the transfer) Maybe a softbutton for blind transfer, and
reserve the hard button for attended transfers? It would be nice if we
could change the behavior of the transfer button to do this.

Conference works, but I have some trouble dropping the correct party from
conversation. Also, when conferenced, I am no longer able to send DTMF
tones to the second line. The original caller hears the DTMF, but the
second party does not. I am using RFC2833 DTMF.

Redial seems to work a bit odd. It does redial a previous number, but not
the last number dialed. Can't seem to find any rhyme or reason, but it
seems do dial the last non PSTN number. For instance, if I dial "50" for
voicemail, then dial 93934481, hang up, and press redial, it dials "50",
not 93934481.

When viewing the SIP config through the menu on the phone, it displays
defaults, and not the settings specified via the TFTP configs.

If I set "qualify=1000" in my sip.conf, Asterisk will send a "poke" to the
phone every few seconds, to make sure it is still alive. If I enable this
option, the phone stops working after a few minutes. Asterisk shows the
phone as unreachable, and I cannot dial any number from the phone. It will
accept the input, but the dialplan does not timeout and dial, nor will it
dial if I press the # sign.

On the subject of dialplans, I am only able to dial 10 digits, on the 11th
digit, the phone tries to dial. This is a bad thing when trying to dial
long distance. It is basically impossible.

The display occasionally shows "L1" in the lower left hand corner of the
display. As if someone had pressed the left/right arrows of the navigation
pad. It shows up for less than a second, and then goes away.

On to the good things now. ;-)

Conference basically works, save for the caviats above.

The backlight is very nice, and the surface of the LCD does not show
fingerprints like the Cisco phones do.

Sound quality is good with ULAW codec, but there is noticable echo in the
call that is not present when using a Cisco phone, or a Softphone.

Transfer seems to work fine.

Message lamp and stutter dialtone works fine. I would like to have a way
to turn off the stutter tone. If I have a message lamp, there is no need
for a stutter tone.

The web config is nice, but limited. I would like to be able to set up SIP
configs through there. Also it only works from Internet Explorer on
Windows. If you are running Netscape/Mozilla/Opera, you cannot use the web
config. It should also be fully password protected, so rouge users cannot
reboot the phone etc.

Custom ringers are nice, and professional, recognize them all from Nortel
phones. I would like to be able to specify a SIP Alert_Info so that I can
have a different ring from different calling partys. For instance,
incoming calls from the PSTN would have one ring, and extension to
extension calls would have another ring.

Look forward to hearing back from you. Nice work on this phone. If it hits
the market with a complete firmware, at $200 or less, they will sell well
within the Asterisk community.

Jeremy Parr
Senior Engineer, Network Services
BGC Ltd.