[asterisk-users] SIP phone recommendation (used to be: no subject)

[Dave Fullerton]

Michael Graves wrote:
> On Mon, 29 Oct 2007 15:01:38 -0400, lists at infoway.net wrote:
> 
>> Well, just general office use. They are a real-state construction  
>> company, so the phones will get some heavy use since most of the  
>> phones are going to sales associates.
>>
>> Now, one of the things we are most interested in are:
>> 1) Asterisk compatibility
>> 2) Mass provisioning
>> 3) Remote management
>> 4) Excellent audio quality (I know there are many factors involved,  
>> but would like to rule out the phone set itself)
>> 5) Robustness
>> 6) Vendor reputation and warranties
>>
>> We have used Linksys 941s in the past and think they're pretty good.  
>> However, we've only used them in 3-5 phones office environments.  
>> We've also used the Polycoms IP 501 and 650s. They seem good, but  
>> sometimes the users complain about the audio being a bit weird in the  
>> sense that, probably, the silence detection may give the user a  
>> feeling that the line dropped. Then again, we've only used these once  
>> (one client installation for each), so for practical purposes, we  
>> don't really have any larger quantity real-life experience.
> 
> For my money it's Polycom every time. It's great hardware. Meets all
> your requirements. 

Granted I have only used Polycom phones, but I would second that vote. 
My experience with provisioning is that it isn't necessarily hard but it 
can be time consuming. Your best bet is to get your firmware extracted 
and then go through the sip.cfg line by line with the admin guide handy 
and tweak as you go. Then repeat with the default phone.cfg file. I use 
a shell script (which I'll share with anyone who wants it) that makes 
adding additional phones a snap. I pass it the name of the default 
template file, the extension number and the MAC address of the phone and 
it creates the MAC.cfg and phone{extension}.cfg files.

> I thought that silence supression was specifically disallowed with
> Asterisk? Something about timing requirements not being met.

I can't say for certain, but that may not be true any more. I came 
across a setting called "internal_timing" that may allow for the use of 
silence suppression. If anyone can comment on that I'd be interested to 
hear what that setting does. This is what I found from Google:

http://forums.digium.com/viewtopic.php?t=15577
http://bugs.digium.com/view.php?id=5374

-Dave