[SATLUG] Since we're talking about Asterisk....

John Pappas j at jvpappas.net
Wed May 21 11:17:03 CDT 2008


As I am walking a similar path, (maybe VoIP for nect SATLUG Meeting?)
help me understand your config, and I will go through mine after that.

On Wed, May 21, 2008 at 12:41 AM, FIRESTORM_v1 <firestorm.v1 at gmail.com> wrote:
> Well....
>
> You're going to LOVE this.
>
> I got the cellpipe to initialize!  All 8 ports even. They show up as x201-x208.
>
> I used freedigits (defunct but the #s still work?) for inbound and
> voipcheap for outbound. (The outbound is limited to 1 minute
> increments, PERFECT for testing!).  On the softphone (x200) incoming
> and outgoing works, albeit choppy because of the VM that it's running
> in.

The CellPipe card is providing FXS (Station) connectivity for the
extensions and your trunk is a freedigit SIP account?  You also have a
SIP extension (x200) configured that is basically a DID setup for the
freedigits number?

If I am understanding correctly, you have the trunk properly set up
(extensions can call out) but the timeout on the FXS ports is really
long (or infinite) so that the system is not interpreting completion
of the station commands without a hint (the #).

> All 8 analog ports can dial out but can't receive calls, and sip show
> peers shows unreachable, which means that I can't call the 8 analog ports

Since the inbound ring is being directed to x200, that indicates that
you need to configure a more detailed inbound route plan, since I am
assuming you want to share the one trunk number with multiple exts.  I
belive that a queue configuration for the inbound call route can be
configured to ring all exts, or in order, or whatever.  That
implementation I have not done by hand as the Fonality GUI I use has
an auto-answer tab that is very easy to configure for complex call
handling rules (time based, caller based, DID based, IVR based, etc).

> With the testing that I've done, I have found a few quirks, so don't
> be looking for a cellpipe as a quick ATA just yet...
>
> 1: Have to dial a # after the number to get it to commit. (Ex, I
> changed the voicemail on the system to 699. In order to get it, I have
> to dial 699# to get it. Outcalls work great, just have to dial
> 912101234567# and the dialing rules strip out the 9 and the #
> 2: Still can't figure out the extension-to-extension calling.  All
> analog ports can dial x200 fine, but x200 dialing the analogs gets
> "Service Unavailable".

I am playing with Trixbox CE (Based on FreePBX/*) without any fancy
hardware (FXO/S or PRI) so I cannot help too much with the routing
configuration required to route calls from SIP to FXS.  All that I can
say is that the FXS ports should have a route in place so that a user
dialing 20x will get routed appropriately.  What happens between 2 SIP
exts?  It would seem that once you set up the routes for the FXS
extenions, you should resolve both the ext->ext issue as well as the
inbound call -> ext issue.

I have found that the going rate for a public residential phone number
is $3-5/month, and usually additional $$ for inbound calls ($/min or
flat/month).  Outbound tends to be the standard $/min based on
destination. A "For Business" DID tends to be a couple $ higher, and
the flat fees are greater, but the $/min costs are usually the same
(must account for both the greater load and greater revenue client :)

Thanks!
John


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