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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
