[SGVLUG] Asterisk

[Mike Fedyk]

SIP is much like Active FTP, you need to open a range of ports to your 
asterisk server, and tell asterisk to use that port range.  check 
/etc/asterisk/rtp.conf

Before you buy grandstream, let me suggest a low-end linksys phone.  You 
will not regret it, and grandstream is quite bad in its reliability, 
sound quality and quality of firmware releases.  check voip-info.org for 
grandstream for a long list of bugs.  I especially like the SPA-941 and 
SPA-942, but the low-end SPA-901 is nice as well.

http://www.voip-info.org/wiki/view/Linksys
http://www.voip-info.org/wiki/view/Grandstream

I have had mixed success using QoS traffic control on the Debian 2.6.8 
kernel with the two scripts linked below (with surprisingly better 
results with CBQ than HTB).  I would get some degraded sound quality 
unless I reserved a portion of bandwidth for VoIP instead of allowing 
the CBQ or HTB priority to work automatically.  There are of course a 
few more things I need to try, so don't think it can't be done.  It is 
probably a good idea to have a dedicated connection used for voip if 
you're using consumer level connections like DSL or Cable.

http://mikefedyk.com/wondershaper-pkt-size-classes
http://mikefedyk.com/wondershaper-pkt-size-classes-htb

I have had good service with voxee, bad with teliax and terrible with 
terravon.  You should have multiple outgoing providers in case something 
happens, and have a rock solid incoming provider.  Unfortunately, I 
haven't found a provider that is more reliable than my vonage or POTS 
line at my house.  Admittedly I haven't been looking recently, but that 
is where I'm at now.

Hope this helps.

Mike



Don Saxton wrote:
> I have had pretty good success setting up and using asterisk with a 
> fxo card I got from x100p.com Last week I was on the east coast and 
> could use a iax softphone, idefisk from asteriskguru.com
>
> I working with someone in Kona, HI to set up another pbx there. We 
> will connect those pbxs to one in SF through IAX.  I think the first 
> hard phones we will try will be Grandstream. One goal is to export the 
> technology to a bunch of non-profits strung out over the US, Canada 
> and then other parts of the developed world. The point is to improve 
> their communications and ability to work together.
>
> One problem that I haven't dealt with effectively is to choose some 
> sip to pots and pots to sip providers.  Does anyone have any guidance?
>
> Another problem is SIP. It seems to easily connect on 5060 but have 
> problems in the RTP port range. Has someone found a good diagnostic tool?
>
>
>
>
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