[asterisk-ss7] chan_ss7 connected to a telco
harry gaillac
gaillacharry at yahoo.fr
Mon Jul 24 14:33:36 MST 2006
Which is the best digium card for building a SIP/SS7
gateway
what about the latest digium cards with a DSP .
--- Anton <anton.vazir at gmail.com> a écrit :
> Have to mention the audio lost issue, binded with
> network
> jitter. For some situations it makes the channel
> driver
> hardly usable...
>
> On 24 July 2006 18:01, Kai Militzer wrote:
> > Hello Harry,
> >
> > My experiences are as followed:
> >
> > harry gaillac wrote:
> > > SS7 protocols:
> > > MTP layers 1-3
> > > ISUP
> >
> > The MTP-Layer seems to be implented fully. At
> least to
> > Layer 2, not sure about Layer 3, but I don't use
> things
> > as failover, so I cannot tell for sure.
> >
> > ISUP is not completely implemented. The most
> message
> > types/features are there and working, but AFAIK
> some
> > things are not there, but also nothing that I
> would need.
> > When I first tried chan_ss7 in late 2005, there
> wasn't an
> > implemantation of CCR test, but we let that
> implement by
> > a developer and since then all features that we
> need are
> > there.
> >
> > > Basic calls :
> > > inbound-outbound
> >
> > Work as they should. Nothing much to say about
> that.
> >
> > > Supplementary services:
> > > Redirecting number
> >
> > I think it got transmited, but I am not sure, if
> this
> > gets somehow into other channel types like SIP.
> >
> > > CLIP/CLIR/COLP/COLR
> >
> > The calling number is always in ISUP messages.
> Only if
> > one bit is set, you must not show it to the end
> point
> > (the user). Asterisk with chan_ss7 can set this
> bit. I
> > don't know if COLP works. Never testes this.
> >
> > > Network configuration:
> > > SEP functionality
> > > Connection to other SEP or STP
> >
> > Never done anything into that direction, so I
> cannot
> > tell.
> >
> > In overall I can say, that it works stable since
> about
> > four month in a friendly-user test.
> >
> > Regards,
> > Kai
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