RE: SB Error + VOIP routing

["Paul Gale" <groups2@xxxxxxxxxxxxxxxx>]

Gregg,

Not having followed this thread before - what will Asterisk and SB allow
you to do? i.e. what functionality?

I need to get my Asterisk box going again - I was using Asterisk @ Home -
should I try that again or do a native Asterisk install? I have an SPA3000
and really wanted to use Asterisk as a glorified answer machine and longer
term get into VOIP. Asterisk was such a steep learning curve :(

Cheers,

Paul.

> -----Original Message-----
> From: xap_automation@xxxxxxx
> [mailto:xap_automation@xxxxxxx] On
Behalf Of Gregg Liming
> Sent: 09 February 2007 03:56
> To: xap_automation@xxxxxxx
> Subject: Re: [xap_automation] SB Error + VOIP routing
>
> Quoting James Traynor (2/8/07 7:37 PM):
>
> > Gregg,
> > This Switchboard also contains outgoing network selection per
contact
> > per number we talked about. It still needs some work but it
should
> help
> > with asterisk routing.
>
> This looks really good!  A couple of questions/issues though:
>
> 1) Now, I'm seeing an automatic "Disconnecting in 7 seconds"
message as
> soon as the dial goes through.  This appears to be new.  Does axc need
> to send a CTI.cmd w/ a CTI.cmd Offhook message to force it to
continue?
>  FWIW: axc is now using the strategy that we discussed where the VPSTN
> line remains declared free since it can create an indefinite number of
> trunked calls.  I'm wondering if the previous line state change kept
> the
> disconnect message from appearing--especially since the original
> behavior persists for (V)PSTN calls.  I'm fine w/ it behaving this
way,
> if I know how to cause axc to maintain the connection.  Another bit of
> information is that axc necessarily uses it's own tag info (comprised
> of
> the two asterisk channels IDs); so, it won't match the CTI.cmd Dial
> msg's tag.  I mention this in case the call termination message can't
> then be used to properly terminate the call.  If this either is or
> might
> be the case, then do I need to also send a CTI.cmd Onhook msg?  I
> suppose the same somewhat applies to the (V)PSTN dialing as the screen
> never realizes that the call is terminated (the disconnect button
> persists despite the call terminating); however, the call record is
> accurate.
>
> 2) This is truly a "nit", but if initiating a dial from the
call log of
> a number that is now associated w/ a contact, it seems that SB still
> assumes the line type to be PSTN rather than propagate the line type
> that is now selectable in the contact record.  In contrast, dialing
via
> the contact record is perfect.
>
> Thanks for adding this as it's making my use of axc and SB much nicer!
>
> For those that may be following this, I will soon be deploying an axc
> release that leverages these capabilities and makes use of axc/SB much
> better.
>
> Gregg
>
>
>
> Yahoo! Groups Links
>
>
>