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Posted by prox, from Charlotte, on January 24, 2007 at 22:59 local (server) time

After toying around with Asterisk over Christmas, I figured it would be neat to actually make use of the monstrosity I created, and use it for voice.

It was clear from the start that Linux soft phones were in pretty shabby shape.  Kphone was the best I used, but it didn't even handle simple DTMF tones, so logging into voicemail was difficult.  It also didn't seem to handle G.711 too well, and crashed almost every time it tried to initialize an RTP stream using the codec.  X-Lite for win32 seemed a bit better.  Anyway, hard phones were in order.

I ended up picking out a Polycom SoundPoint IP 300 mostly due to a recommendation from Lily, and the fact that the IP 301's replaced the 300's, just recently.  It ended up going for roughly $80 on eBay.  Aside from a funky web interface (that requires a complete reboot for any configuration change) and lack of two-way speakerphone, it works quite well.

SoundPoint IP 300

Configuration of Asterisk was pretty simple, due to Debian's nicely-commented configuration files and frequent visits to voip-info.org.  In almost no time I had a couple extensions configured, voicemail working, and some trancy hold music.  It became apparent that some NAPTR and SRV records were required for a complete SIP setup, and I modified DNS settings after reading about the proper way of doing it.  I still think there might be a misconfiguration or something, since I still have to specify the SIP server name, in addition to just the URL.

So, I picked up two more 300's, and sent them to my parents and grandparents.  The project seems to have been a success, at this point.  I was going to setup some QoS, but it didn't really seem to be needed.  Call quality is still crystal-clear, even when maxing up the upstream on my residential Road Runner connection.  Asterisk also supports reinviting, so a call from NJ to Florida, although negotiated via my Asterisk server in Charlotte, is established directly between the two phones.

Now, I'm looking around for some [free] public SIP termination services.  It'd be neat to link up this system to the PSTN, even if it's for outgoing calls only.  I have a feeling I might have to pay, though.

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