Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, Christophorus
Philipp von Klitzing
2007-Jul-14 12:26 UTC
[asterisk-users] Zaptel/mISDN and call transfer
Hi!> I am searching for a possibility to do a certain call transfer method > which is called "path replacement" in QSIG. But I want to do that in > DSS1 (EuroISDN).They keyword to search for is "explicit call transfer" (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes with support for that. Don't know about mISDN. Cheers, Philipp
Philipp von Klitzing
2007-Jul-14 17:07 UTC
[asterisk-users] Zaptel/mISDN and call transfer
Hi!>>> I am searching for a possibility to do a certain call transfer method >>> which is called "path replacement" in QSIG. But I want to do that in >>> DSS1 (EuroISDN). >>> >> They keyword to search for is "explicit call transfer" (ECT). At least >> chan_capi-com (http://www.melware.org/ChanCapi) comes with support >> for that. Don't know about mISDN. >> > Thanks, but can I use chan_capi as frontend to mISDN or zaptel > hardware?You can run chan_capi on top of mISDN, but I have never done this myself (be careful to not confuse mISDN with chan_misdn). Anyway, I am not sure what "mISDN hardware" would be, whereas "CAPI hardware" would refer to cards that come along with a CAPI interface/driver like Eicon Diva or various AVM products, possibly also HST products.> As I know I do have to choose between digium or beronet/junghanns > hardware (E1) to use PRI with asterisk, right?Or Sangoma (which uses zaptel), or Sirrix (comes with its own channel driver). My personal suggestion would be that you take a closer look at Eicon and Sangoma.> Do I have to use chan_capi to access the zaptel hardware?That won't work. By the way, you might want to also search for "call deflection" (CD) and "partial reroute" (for PtP = point-to-point connections = "Anlagenanschluss") next to ECT. See: http://www.voip-info.org/wiki/view/ISDN+Features http://www.voip-info.org/wiki/view/Asterisk+CAPI+readme http://www.melware.org/ChanCapiCallReroute Cheers, Philipp
Tzafrir Cohen schrieb:> On Sat, Jul 14, 2007 at 01:23:35PM +0200, Christophorus Laube wrote: > >> Hi list, >> >> I am searching for a possibility to do a certain call transfer method >> which is called "path replacement" in QSIG. But I want to do that in >> DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine >> to signalize on dchan that the call path has to be replaced to a direct >> connect between the caller and the called, i.e. my machine is to hang up >> after the transfer and the channels are free again. Is it possible and >> with what card vendor (mISDN vs.zaptel) and how do I do that? >> Thanks in advance, >> > > I found an old feature-request bug in Zaptel which seems relevant: > > http://bugs.digium.com/3554 > > Not sure if this means that the feature is supported. Maybe ask Mathew > Fredrikson or Digium support. > >by the way: Is this call deflection or ECT etc. only possible to be executed at ring time or can I redirect a yet running call? Thanks, Christophorus
Philipp von Klitzing
2007-Jul-15 21:57 UTC
[asterisk-users] Zaptel/mISDN and call transfer
Hi!>> I found an old feature-request bug in Zaptel which seems relevant: >> http://bugs.digium.com/3554 >> Not sure if this means that the feature is supported. Maybe ask Mathew >> Fredrikson or Digium support. > > by the way: Is this call deflection or ECT etc. only possible to be > executed at ring time or can I redirect a yet running call?ECT only works after having answered a call (and having put it on hold on the ISDN side, not in Asterisk), see readme of chan_capi-cm for more info. Deflection, on the other hand, must be arranged before a call is answered. I add to the wiki a tiny overview for the different methods and channels: http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer Of interest for you are probably: * chan_misdn has this dialplan app: misdn_facility(calldeflect|9999) * ZapCD for deflection as implemented by bristuff Cheers, Philipp