asterisk users - Dec 2005 - Does hardware like this exist...?

Hi all!

I am looking for a device that I can stick in a USB-port on my Asterisk server
and that allows me to connect one/more (cordless) PSTN-phones in such  a way
that they'll work with SIP/Asterisk. I know
there are USB-phones, but what I'm looking for is 'the USB-phone without
the phone', if you know what I mean...   ;-)


Regards,
	Evert

On 12/16/05, Evert Meulie <evert@witelcom.net>
wrote:
> Hi all!
>
> I am looking for a device that I can stick in a USB-port on my Asterisk
server and that allows me to connect one/more (cordless) PSTN-phones in such  a
way that they'll work with SIP/Asterisk. I know
> there are USB-phones, but what I'm looking for is 'the USB-phone
without the phone', if you know what I mean...   ;-)
>
 You're looking for a USB FXS port. Yes, they do exist. You can take a
look at the Astribank-8 from Xorcom (www.xorcom.com). I really don't
know how well they work as I haven't any personal experience with
their equipment, but they were exhibiting this solution at the last
Astricon a few months back.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

All,

I have the following set up:

    Fedora Core 4 box (yum updated to current)
    Asterisk 1.2.1 + Chan_Capi-cm-0.6.1
    AVM C4 card
    2 x ISDN2e lines bonded with switchboard number, fax number and 10 x DDI 
numbers from British Telecom
    14 x Cisco 7960 phones with SIP 7.5

The ISDN lines work in P2P mode and calls are presented with the last 4 
digits only - I land them in a context and branch out from there - 
everything to do with incoming calls works just fine!

I have a problem with outgoing calls that are routed over the BT network and 
the way in which 'ringing' is presented... depending on the called party
number (hence phone provider) I get different results. For example:

a) if I dial another BT number I get a fraction of a second's ring followed 
by silence until the called party answers. The Cisco phone displays:

    Proceeding (in 100)

very briefly and is almost immediately over-written by:

    Session Progress (in 183)

until the called party answers - at no point is Ringing Destination (in 180) 
displayed


b) if I dial an Orange or O2 mobile number I get a second or two's worrth of
silence [while the Orange network locates the mobile] then the mobile rings 
in the normal way and the Cisco phone plays out US style ringing. When the 
number is dialled the phone displays:

    Proceeding (in 100)

when the mobile starts to ring the Cisco phone displays:

    Ringng Destination (in 180)


c) if I dial a Bulldog phone number then I get three messages:

    Proceeding (in 100)  - for a second or so
    Session Progress (in 183) - for a couple of seconds
    Ringng Destination (in 180) - while the called party's phone rings


d) and the really weird one - if I dial *some* international numbers I get 
both UK (BT) ringing tone overlaid with Asterisk/VoIP (US) ringing tone



I have two ways of dialling out:

1. with an explicit "9" for an outside line -- get dialtone from BT
and then
dial rest of the digits - like a legacy PBX

2. dialing just based on the fact that the extension starts with a zero so 
its an outside call via BT


I have tried all combinations of early B3 connect 'always', 'on
success' and
'never' and it doesn't appear to change things... the relevant part
of
extensions.conf is below for completness.

Before I dive in to the next level down:

- is this a known issue?
- is there a solutiuon/workaround/patch/fix
- do I need to get down and dirty with CAPI and SIP debug?


Mike




;
; external-routes: this is where we get to dial out
;
[external-routes]

;
; outgoing via main ISDN line using explicit "9" for an outside line
; and ISDN eqarly B3 connect ("overlap sending") to drop us to the
; BT provided dialtone and work like a normal/legacy phone system -
; we force the caller ID to our exchange number so that DDI's dont
; leak out
;
exten => 9,1,NoOp("ISDN: Pickup outside line (early B3 connect) for: 
${CALLERIDNUM}")
exten => 9,2,SetCallerId(${THORCOM_MAIN})
exten => 9,3,Dial(CAPI/g1//b)
exten => 9,4,Hangup

;
; implicit trunked call - here we could/should do an ENUM look
; up to see if we can place the call via IP and fall back to BT
; if not... just for now this isn't implemented and we always call
; out via BT!!
;
exten => _0.,1,Dial(CAPI/g1/${EXTEN}/b)                    ; early B3 
connect always
;exten => _0.,1,Dial(CAPI/g1/${EXTEN}/B)                   ; early B3 
connect on success
;exten => _0.,1,Dial(CAPI/g1/${EXTEN})                       ; no special 
options
exten => _0.,2,Hangup