asterisk users - Feb 2004 - Comments on Voice Quality IP Hard Phones

Good day,

I am in the middle of getting my self some hard phones. Anyone care to 
comment on the *voice* quality of the following phones:

Cisco 7960
Siptone II
SNOM
Budgetone

I have seen a few reviews, but none go to deep into the voice quality 
issue.

Thanks.

rgds
pos

On Wed, 25 Feb 2004, Peer Oliver schmidt wrote:
> Budgetone
>
> I have seen a few reviews, but none go to deep into the voice quality
> issue.
I don't mind the voice quality, I just wish it would always be working
when I picked up the handset.  Mine tend to lock up (they are behind a
firewall, I am running current CVS, have upgraded firmware, etc).  If they
fixed their software (maybe even opened it up for others to write), I
think it would be a heck of a deal for a phone.

-- 
Joel

> I am in the middle of getting my self some hard phones. Anyone care to 
> comment on the *voice* quality of the following phones:
> 
> Cisco 7960
> Siptone II
> SNOM
> Budgetone
> 
> I have seen a few reviews, but none go to deep into the voice quality 
> issue.
In theory (and mostly in practice), each of the phones are going to provide
roughly the same voice quality "if" each has been configured to use
the
same codec (ulaw as an example). There are lots of other considerations
that might weight heavily towards which phone is more usable/practical
under certain conditions.

For example, the 7960 is a very solid business-class phone that non-technical
users can relate to. Speakerphone functions very well, instrument is heavier
and doesn't pull across the desk like some of the light-weight ones do, 
front panel buttons are intuitive, the sip protocol is very reliable, etc.

The Snom 200 phone mostly functions well, however the phone's logic is more
oriented to european telephony and several of the functions do not work in
a manner that one might consider 'standard' in the US. It's
light-weight,
pulls across the desk when the handset cord is stretched, handset is 
somewhat different when compared to old analog phones, some of the front
panel keys have multiple functions (mostly undocumented) that can get one
into trouble, etc. Still a good phone with many good characteristics
including voice quality, but if you let a non-technical user eval both the
Snom and Cisco, the user will pick the Cisco just about every time.

I've not used the Siptone II or Budgetone, but if memory serves correctly,
those two are more entry-level low-cost devices that were targeted at a
different market then were the 7960 and (at least some) Snom phones.
Can't speak to their quality at all.

Seems each of those manufacturers have at least "some" issues
remaining to
be resolved in terms of user functionality, technical compatibility/problems,
features, exposure to security issues, reliability, etc.

At 05:26 AM 2/25/2004, you wrote:
>I am in the middle of getting my self some hard phones. Anyone care to
>comment on the *voice* quality of the following phones:
>
>Cisco 7960
>Siptone II
>SNOM
>Budgetone
>
>I have seen a few reviews, but none go to deep into the voice quality
>issue.

I have not received any complaints here about the voice quality on 13 SNOM 
phones that I installed a few days ago. There have been several complaints 
about other things (mostly unrelated to the phones themselves), so I'm sure 
I would have heard :) We are using the g711a codec in-house.

--Ernest


>Thanks.
>
>rgds
>pos
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>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
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>    http://lists.digium.com/mailman/listinfo/asterisk-users

On Wednesday 25 February 2004 08:26 am, Peer Oliver schmidt
wrote:
> Good day,
>
> I am in the middle of getting my self some hard phones. Anyone care
> to comment on the *voice* quality of the following phones:
>
> Cisco 7960
> Siptone II
> SNOM
> Budgetone
>
> I have seen a few reviews, but none go to deep into the voice quality
> issue.
>
> Thanks.
>
> rgds
> pos
I have Budgetone and the speaker phone is not the best but the handset 
quality is quite OK. It does not support the "full" audio spectrum of
a
phone so a deep voice will not come out quite as deep. Other wise it's 
waay good for a $80 VoIP speakerphone.

Actually, when I make a VoIP call coast to coast it's so quiet I cannot 
tell if the connection is up, but that's thanks to *.
-- 

"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
                                Benjamin Franklin

I've got several of the Polycom IP 500's as well.  I was using
snom 200's.      They were horrible speakerphones and the echo inside
the handset made it     almost unusable.  The 100 was completely
unusable.  For less than $60 per     phone more, I got a phone that I
have to spend less than half the time     working on.  Configuration is
completely config file controlled as opposed     to controlling the
phone from a webserver built into it.Chris Tooley
>>>mwless@mindworks.net 02/25 12:57 pm >>>
We implemented 21 Polycom IP 500 SIP phones in December and the voice
    quality with Ulaw is very good. We also use g729 and IAX to connect
two
sites together on DSL and voice quality sounds pretty good as well. No
one has complained yet.

The speaker phone is pretty good, and     occasionally on POTS calls
when
using the Handset we have echo for the     duration of the call. This
however, seems to be an issue with *     and not the IP 500's.

We also tell the receptionist to reboot her     phone daily, as the
transfer
feature will stop working occasionally,     but the reboot has remedied
the
problem.

Rgds,

Brent
>-----Original Message-----
>From: asterisk-users-admin@lists.digium.com     [mailto:asterisk-users-
>admin@lists.digium.com] On Behalf Of Jonathan     Moore
>Sent: Wednesday, February 25, 2004 1:46 PM
>To:     asterisk-users@lists.digium.com
>Subject: Re: [Asterisk-Users] Comments     on Voice Quality IP Hard
Phones
>
>Want to add a little something     about the BG speakerphone. It is not
very
>loud,
>but it is     pretty good other than that. I have tested snom 100, snom
200,
>and
>siptones and BG is the only one that was remotely close to usable. The
    bg
>actually has a full duplex speakerphone which is not all the common    
in
a
>lower
>cost unit. It is probably usable for a single     person in a fairly
quiet
>environment. I have the snom almost working     with the speaker but
since
it
>is
>half duplex and I have an     ajoining server room which creates a lot
of
>background noise the     remote end sound pops in and out.
>
>For a conferening phone     module I am looking at using and analog
Polycom
>SoundStation with a     sipura IP converter.
>
>
>--
>Jonathan Moore
    >Director of Technology
>Winfield Public Schools
>Office     620.221.5100
>Fax 620.221.0508
>
>
>Quoting Steve     <steve@szmidt.org>:
>
>>On Wednesday 25 February 2004 08:26     am, Peer Oliver schmidt wrote:
>>>Good day,
>>>
    >>>I am in the middle of getting my self some hard phones. Anyone
care
>>>to comment on the *voice* quality of the following phones:
>>>
>>>Cisco 7960
>>>Siptone II
>>>SNOM
>>>Budgetone
>>>
>>>I have seen a few reviews, but     none go to deep into the voice
quality
>>>issue.
>>>
>>>Thanks.
>>>
>>>rgds
>>>pos
    >>
>>I have Budgetone and the speaker phone is not the best but     the
handset
>>quality is quite OK. It does not support the full     audio spectrum
of a
>>phone so a deep voice will not come out quite     as deep. Other wise
it's
>>waay good for a $80 VoIP speakerphone.
>>
>>Actually, when I make a VoIP call coast to coast it's so quiet I
cannot
>>tell if the connection is up, but that's thanks to *.
>>--
>>
>>They that would give up essential liberty for temporary safety
deserve
>>neither liberty nor safety.
    >>Benjamin Franklin
>>
>>_______________________________________________
    >>Asterisk-Users mailing list
>>Asterisk-Users@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To     UNSUBSCRIBE or update options visit:
   
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>Visit Winfield Public Schools at http://usd465.com
>-------------------------------------------------
>This mail sent     through IMP: http://horde.org/imp/
    >_______________________________________________
>Asterisk-Users     mailing list
>Asterisk-Users@lists.digium.com
   
>http://lists.digium.com/mailman/listinfo/asterisk-users
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At 08:26 AM 2/25/2004, you wrote:
>Ernest,
>
>Wondering if I could could get some feedback about your system and how
it's
>performing, as we are also considering replacing our existing pbx with * ...
>How many phones do you have total using * .. 13 ? How many co lines ? pri ?
>Are you using a long distance provider like nufone, etc ? How has your long
>distance experience been ? What complaints are you getting ? Have you been
>able to solve these ?
2/25/2004

I've been planning to post to the list about this, so I'll go ahead and 
reply to this personal email...

We are using 12 phones in the office with one spare. All are SNOM 200 model 
phones (more about this later) that we purchased from ABP Technology 
(http://www.abptech.com/). We have a total of seven co lines, all analog 
(plus two DS3 for our Internet connection :) right now, that we connect 
using an Audiocodes MP-108FXO gateway (Also from ABP). The first four lines 
are part of a PacBell hunt group, while the next two are reserved for 
outgoing calls. The last line is a private incoming line to our Network 
Operations Center.

We have the MP108 configured to use the first six lines for outgoing calls, 
but to give priority to the last two, keeping our incoming lines clear. 
Incoming calls cause our receptionist phones to ring for 30 seconds, then 
play a message ("all our staff are busy...") and put the caller into
an
Asterisk Queue. We are eagerly awaiting the upcoming patch to announce 
queue position while in the queue. At any time callers can dial the 
extensions for Sales (1000), Customer Service (1001), Customer Service 
(1002) or Technical support (1003) all of which ring at several different 
phones. They can also dial direct extensions for any of our salespersons.

Some of our direct dial extensions are available to customers, others can 
only be used during a transfer from a receptionist. This protects our CEO 
and Senior Tech Support people from annoyances. Similarly, not all 
extensions will go to voicemail... Customer Service and Tech Support do not 
- they play a message to call back later or send an email.

The MP-108:
We were recommended this product by ABP and have been quite happy with it. 
The documentation is pretty weak, so I spent a couple of hours learning how 
the features work. The 4.0 build that came on the hardware is useless, so I 
immediately upgraded to 4.2 which solved all my problems and added some 
features I hadn't known I needed.

The 4.2 build supports polarity reversal, silence detection and current 
drop for disconnect detection. We use current drop here, which has the 
annoying tendency to take a couple of seconds to notice that someone has 
disconnected. I don't know whether this is an Asterisk thing, a telco 
thing, or a MP-108 thing, but I don't remember having any trouble with 
Digium's FXO card set to kewlstart.

Thanks to Clif Jones for a sample INI file and some advice in getting the 
MP-108 to work.

The SNOM 200 phones:
Audio quality and latency have not been an issue, which is to say that I've 
received no complaints. Connecting them to Asterisk is no trouble at all, 
though I'm having some problems getting them to stay registered (I think 
this is my fault, but I haven't had the time to debug it). I find the 
CallerID display to be very useful. And, once we got used to it, the 
conference calling feature became one of our favorite features.

The handsets are on the light side and the handset cords are just a little 
too short for large desks, but this is a problem that can happen with any 
phone, and they do use standard RJ-12 connectors. The headset and 
Speakerphone features are extremely high quality and well thought out - in 
general - and work as well as one could hope. You can't (quickly - and by 
this I mean in less than a second while the phone is ringing) change 
between headset and handset until AFTER you answer the call, but this is 
not a problem in most situations.

Our only major problem with the phones right now is with their surprise 
conference call feature. Here's how it works: If you have one call on the 
phone, you can press the transfer button to transfer, and hang up the 
handset to hang up. Great, just like a normal phone. Now the catch: If 
there is a call on hold and you want to hang up on the person you are 
talking to... DON'T HANG UP THE HANDSET. Doing so will conference the live 
call with the call on hold. OUCH! Similarly, don't use the transfer button, 
because this conferences the call on hold into the live call.

So far, I don't know of any way to do an unattended transfer while you have 
someone on hold. There are workarounds for most of this, but the sheer 
inconsistency of it is driving our office manager up the wall. Now, I may 
still have some settings wrong, but I think I've been pretty thorough. ABP 
is being very helpful in getting this figured out.

My solution: Asterisk has the ability to intercept the # key and use it as 
a transfer button. We've instructed our staff to a) never hang up by 
putting down the handset, and b) never use the SNOM's built-in transfer 
button. Instead, they press ESC to hang up and use the # key to initiate a 
transfer.

ABP also tried to sell us a Power-over-Ethernet device that would provide 
power to the SNOM phones during a blackout (in conjunction with a UPS of 
course). This is a problem that affects all VoIP phones, but our wiring is 
not PoE friendly. So, we went with the external power supplies for an extra 
charge. When we recommend/resell SNOM phones to our customers, we intend to 
sell the PoE system as well, wiring permitting.

All-in-all the office staff here is settling down to the new phones and 
there have been no show-stopping issues so far.

Asterisk:
Wonderful, it does everything I want. A few of the things it does:
- Voicemail
- Private and public extensions
- Directory Service
- Call Queues
- Music on Hold (donated by our local High School jazz band)
- Works with Cisco, SNOM, Pingtel, Budgetone, X-Lite and more
- Automated attendent and IVR
- A simple app that I wrote to play a message based on our network status 
(we are an ISP)
- Fully customizable ring groups (i.e. ring all phones after hours, or only 
the receptionist during hours)
- Conference Calling and Three-way-calling (enhanced by the SNOM phone's 
conference call feature)
- Automatic Failover from PSTN to Internet and vice-versa
- Support for multiple VoIP dialtone providers for low-cost long-distance
- Transfer to cell phone

VoicePulse:
We use VoicePulse for our outgoing long-distance, and so far have not had 
any complaints. We were using NuFone, but are turned off by their lack of a 
web interface for refilling our account and viewing CDR. Both NuFone and 
VoicePulse have recently had (reported on the list, but not personally 
confirmed) outages that did not affect us. In both cases, setup and 
ordering were quick and easy (more so with VoicePulse).

ABP Technology Partners:
After an initial testing period with Asterisk and X-Ten, we purchased a 
single SNOM phone from ABP (I don't recall how we found them). They were 
very helpful and were willing to sell us a single phone, which is always a 
pleasant surprise when dealing with distributors. The single SNOM phone we 
received worked well, so we went ahead and purchased an additional 12 
phones and the MP-108 gateway on the recommendation of our salesperson. Our 
experience with them has been very good, and the (uncompleted) RMA process 
of one defective unit has gone smoothly so far.

--Ernest W. Lessenger
OACYS Technology

OACYS TECHNOLOGY is a 23-year company, founded in 1982 to develop and 
deploy computer solutions. Based in a semi-rural community, the company has 
long been accustomed to operating independently and developing self-reliant 
solutions with minimal external dependencies. We have developed our own 
solutions to address and manage substantial competition and adversity, and 
we continue to do so in the course of our own daily operations. For our 
consulting clients we explain step-by-step what we have done, why we have 
done it, and how to do what we did (or would do or avoid) to resolve any of 
the many challenges facing today's independent ISP/WISPS who are interested 
in winning their own battles and wars.

http://www.oacys.com/ 
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