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

FIRESTORM_v1 firestorm.v1 at gmail.com
Wed May 21 19:54:31 CDT 2008

Here's how i'm set up and I'm trying to be as concise as possible
because work kicked my posterior today and I'm really exhausted. :P

On the cellpipe, Voice channel set to SIP. Sip options set to point to
asterisk with first port starting at 201 There's not much
configuration after that on the cellpipe.

On Asterisk,
created 8 extensions (201-208) and checked for dialtone on all ports.
Also checked with code 601 (Announce Extension) to verify connectivity
to server.

The freedigits trunk is inbound only, and it is set to ring 200 right
this moment because of an issue with using 600 (ring all phones). When
it attempts to ring all, it goes immediately into the "Unavailable"
error message and I get tri-tone with the recording. With it set
straight to any of the extensions on the cellpipe, I get the same
error.  Set it to 200 and it works like it should.

The voipfree trunk is outbound only and there are no inbound routes
defined for it.  This has been verified working from all phones in

It would appear that something's up on the Cellpipe as two softphones
ring as expected and all is normal with the exception of the
nonfunctioning 8 analog ports.

The Softphone is set up with Ext 200 and works fine as is.  Since the
Cellipipe is all SIP, I don't think that the dialingplan for the FXO/S
ports are required, as they are handled like any other (batch of)
softphones, however I could be wrong.

I think Asterisk doesn't like Lucent but then it could be the other way around.


On Wed, May 21, 2008 at 11:17 AM, John Pappas <j at jvpappas.net> wrote:
> 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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