On Sat, 24 Dec 2005, Michael J. Tubby G8TIC wrote: > I changed the dial-string to include flags 'ob' as you mentioned (below) > and now I get the following when I dial a BT phone number > > - dial number, get: > > Proceeding (in 100) briefly > > - after a second or so: > > Ringng Destination (in 180) > > - double ringing tone: > > BT style ringing generated by the exhange > Cisco phone US-style ringing (generated by the phone) > > these are overlaid on each other (mixed together) > > > My hunch is that there's something not right with the call set up sequence > and CAPI handling.
This is not a problem of CAPI. When you specify 'b' for early-b3, you will get the tones from the switch. If your phone adds its own tone, even when it receives progress tones, then it is incorrect (maybe wrong setup). Armin > I'll send you some protocol traces off list. > > Regards > > > Mike > > > > > ----- Original Message ----- From: "Armin Schindler" <[EMAIL PROTECTED]> > To: "Michael J. Tubby B.Sc (Hons) G8TIC" <[EMAIL PROTECTED]> > Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Sunday, December 18, 2005 3:12 PM > Subject: Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with > ringing > > > > On Fri, 16 Dec 2005, Michael J. Tubby B.Sc (Hons) G8TIC wrote: > > > 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? > > > > Have you tried CAPI-Dial option 'o' ? Together with 'b' it should give > > you progress in any case. > > > > Armin > > > > > 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 > > > > > > > _______________________________________________ --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
