[SGVLUG] Asterisk

[Don Saxton]

Got sip working (it was problem on other end). Looking at spa941 ($130 
which is at least 2600 att minutes). Will try voxee and gizmo which is 
slightly cheaper. Incoming can be direct to my pbx. I expect to be in 
the market for DID, so will start with gizmo dial in.  Don't know 
anything about qos measurement or control. The largest office I know of 
has 7 phones. What are CBQ an HTB?

I would like to add a iax or sip client to my smartphone as a second to 
last node on my followme. Last being the cell phone itself. Installed 
microsoft portrait, but haven't yet got a msn acct (I am real 
resistant). It acts flakey and pops up without being asked. I am hoping 
that truphone or somebody will soon have an alternative. There are 
several for pocket pc and symbian.  Good niche for a java phone, but 
only promises yet.

Good to know there is a asterisk presence here.

Mike Fedyk wrote:
> 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?
>>
>>
>>
>>
>>
>
>