asterisk users - Mar 2004 - Opinion poll: best SIP phones for asterisk?

Hi All-

I've just finished setting up my first asterisk box with an X100P and 
TDM40B, with basic asterisk config mostly from ManxPower's sample 
configs.  Thanks for posting those, Manx, and thanks for your help on 
IRC.

I'm also in the midst of carefully studying John Todd's config files, 
and I'm pretty sure that I'll be using large chunks of his in my own.  
Thanks in advance for sharing those, John, and for such extensive 
comments and documentation in your files.

As I'm doing this, I'm considering installing an asterisk box at my 
office (about 6-10 different phone stations) and would like to get 
opinions on the best quality and/or most well-supported SIP hard phones 
and SIP soft phone clients.  I see from John Todd's config files that 
the ATA-186 has (or had) some problems with asterisk, and I'm guessing 
that there are probably some other flukes that crop up with certain 
phones being connected to asterisk, so I thought I'd ask here before 
purchasing something in hopes of avoiding unnecessary hassles.

Many thanks to Mark Spencer and the rest of the development team for 
creating and sharing asterisk.  It's a very nice product.  I've been 
wanting something like this for a long time, and it sure looks like 
asterisk does now (or will soon do) all that I've been wanting and 
plenty more.  Though I'll admit to suffering some major frustration 
from a scarcity of documentation, I think I may finally be starting to 
get it through looking at examples and pointers from folks in IRC.

-Kevin

PS. What's the best source of documentation for the extensions.conf 
file?  I don't see any in the second draft of the Asterisk Handbook, 
not much that I can really make sense of in the wiki page 
(http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf) 
which seems to be full of forward references and many apparently 
asterisk-specific references without definitions, and I don't see 
anything specific to extensions.conf on 
http://www.digium.com/index.php?menu=documentation.  So far, the most 
useful documentation of that file that I've found is John Todd's config 
file with extensive comments.  Is there anything more complete?  If 
not, just knowing that this is the case will help me because I'll stop 
hunting for something that's not there.  TIA.

> PS. What's the best source of documentation for the extensions.conf 
> file?  I don't see any in the second draft of the Asterisk Handbook, 
> not much that I can really make sense of in the wiki page 
>
(http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf)
> which seems to be full of forward references and many apparently 
> asterisk-specific references without definitions, and I don't see 
> anything specific to extensions.conf on 
> http://www.digium.com/index.php?menu=documentation.  So far, the most 
> useful documentation of that file that I've found is John Todd's
config
> file with extensive comments.  Is there anything more complete?  If 
> not, just knowing that this is the case will help me because I'll stop 
> hunting for something that's not there.  TIA.
Probably one of the toughest things to learn/understand in the begiinning
is the concept of "context", how its used, etc. As soon as you think
you
understand it, it seems someone comes along with another approach that
will blow your previous understanding. Most everything else is rather
straight forward. The sample config files located in /usr/src/asterisk/configs
include a fair amount of descriptive text. Also from the CLI, 'show
applications'
and 'show application dial' will become your friends.

I don't know of any easy to read text that actually describes
"contexts" and
how they are implemented.

--On Monday, March 29, 2004 8:24 am -0500 Kevin <asterisk@gnosys.biz>
wrote:
> Hi All-
>
> As I'm doing this, I'm considering installing an asterisk box at my
> office (about 6-10 different phone stations) and would like to get
> opinions on the best quality and/or most well-supported SIP hard phones
> and SIP soft phone clients.
Lot's of references to this topic on the list - a quick search will provide 
loads of feedback.  At a top level:

- cheap option - Grandstream Budgetone - works well but current firmware is 
buggy
- best option - Cisco - good * integration.  Make sure you get the phone
  supplied with SIP installed to avoid any support issues
> I see from John Todd's config files that
> the ATA-186 has (or had) some problems with asterisk, and I'm guessing
> that there are probably some other flukes that crop up with certain
> phones being connected to asterisk, so I thought I'd ask here before
> purchasing something in hopes of avoiding unnecessary hassles.
>
- works fine but call transfers can be tricky to action

  Iain

Hi,
> As I'm doing this, I'm considering installing an asterisk box at my
> office (about 6-10 different phone stations) and would like to get
> opinions on the best quality and/or most well-supported SIP hard phones
> and SIP soft phone clients.
I had great luck with sipura spa-2000 adapters. They can make 3 way
conferences and supervised and blind transfers by themselves. My advise is,
whatever you choose, try to use the same brand of phones. Don't mix sipuras
with grandstreams and ciscos. Good luck,

Nicolas

I've had some minor problems with the spa-2000
but once you fixed the issues they are pretty solid unit
to use. so i'd recommend them highly but i can't wait for those new iaxy
units from digium to come out
those will hopefully resolve some of my firewall issues.






-- 
Best regards,
Frankie       (fgravato@cfsdigital.com)                     
mailto:nanog@cfsdigital.com