SLUG Mailing List Archives
Re: [SLUG] Ask SLUG - IP Telephony
- To: slug@xxxxxxxxxxx
- Subject: Re: [SLUG] Ask SLUG - IP Telephony
- From: Daniel Pittman <daniel@xxxxxxxxxxxx>
- Date: Tue, 10 Feb 2009 17:31:08 +1100
- Organization: I know I put it down here, somewhere.
- User-agent: Gnus/5.110006 (No Gnus v0.6) Emacs/23.0.60 (gnu/linux)
Gonzalo Servat <gservat@xxxxxxxxx> writes:
> On Tue, Feb 10, 2009 at 1:54 AM, Jake Anderson <yahoo@xxxxxxxxxxxxxxx>wrote:
>> I have done a few installs with asterisk based soft/hardware.
>> Expect some teething troubles. Get a dedicated adsl line if you are heavy
>> internet users, QOS hasn't been much of a problem for most installs however.
>> Run separate networks for voice and data.
>> I use grandstream for the physical phones they seem pretty good, perhaps a
>> little fragile physically (they reset if you bash them, moral of the story,
>> don't bash them). Do NOT use cisco phones, that way lies madness.
>> don't bother trying to run a hybrid between your new and old systems. Run
>> in parallel for a bit if you can with all outbound calls on the new system.
>> Then dump the old and go full IP.
>> If you want some help with setup and demo drop me a line
> Another suggestion: check out FreeSWITCH. It's an alternative to Asterisk
> and I would personally never go back to Asterisk unless I absolutely had to
> (ie. gun to my head).
Actually, out of curiosity, and since I want to get rid of Asterisk and
replace it with something (anything, so help me, anything at all) else.
FreeSWITCH is popular at the moment; the only other convincing option
I have run across is yate.
So, dear lazyweb, can you tell me:
Have you actually used FreeSWITCH or YATE in a small SIP-only environment?
Did it work well, reliably and with minimal maintenance?
Did it work effectively as an answering machine for home?
(That is, play a message, record the caller message, notify you,
with nothing much more fancy than that available?)