[asterisk-ss7] My experiences with chan_ss7, some questions and a solution for the ringback tone

Fri Mar 17 21:48:31 MST 2006

Leonimar, :) I've been posting that in the mailing list and 
voip-info.org my patch FIXES that - because I did the same 
protocol analysis and I've written about in voip-info - 
that is a difference of the ITU and ANSI ISUP

Look at the erricson presentation saying about that.
http://www.google.com/search?q=von9909_ericsson.ppt&sourceid=mozilla&start=0&start=0&ie=utf-8&oe=utf-8

von9909_ericsson.ppt

On 17 March 2006 20:06, leonimar cape wrote:
> Hi everyone,
>
> Still with regards to the RBT. After attaching a
> protocol analyzer, I found out that asterisk is not
> sending ACM after receiving the IAM from the Nortel
> DMS switch. Also I notice "unknown state 0x03" on the
> channel. What does this mean. Can anyone help. Is this
> message already included in the new release version?
> Nonetheless, I have not problem with calls orginating
> from the asterisk to the pstn side.
>
> Regards,
>
> Leonimar Cape
>
> --- Are <london3 at gmail.com> wrote:
> > Hi
> >
> > I fully support the Timeout parameter as this is a
> > common practice in SIP
> > based communication.
> >
> > I work a lot with Patton SmartNode Sip Gateways and
> > in the configuration we
> > have the following.
> >
> > context cs switch
> >   digit-collection timeout 3
> >   routing-table called-e164 TEST1
> >     route .T dest-interface IF_E1
> >     route 00.% dest-interface IF_E2
> >
> > On many SIP phones you also have the option to
> > choose Timeout or *Early
> > Dial  *(484 response)
> >
> > You are not fully aware of your call routes in many
> > Real life SIP
> > applications. We all know that International
> > numbering plans are no easy
> > beasts.
> >
> > --
> > Are Casilla
> > http://astartelecom.com - Independent VOIP Telecoms
> > Broker. Asterisk
> > Consultants
> > http://astbill.com - Open Source Billing, Routing
> > and Management software
> > for Asterisk and VOIP
> > AstBill DEMO: http://demo.astbill.com
> >
> > On 3/16/06, Kai Militzer <km at westend.com> wrote:
> > > Hello Jacob, hello all,
> > >
> > > Jacob Tinning wrote:
> > > > We didn't like the timer-solution because we
> >
> > think its wrong to delay
> >
> > > all calls
> > >
> > > > X seconds just because the SS7-asterisk doesn't
> >
> > know another Asterisk's
> >
> > > dialplan.
> > >
> > > Thats why I made it configurable, so that it can
> >
> > be turned off, when not
> >
> > > needed. ;)
> > >
> > > > My suggestions is
> > > >  1. Use identical dialplans on the SS7-gateway
> >
> > and the SIP server
> >
> > > >  2. Store the dialplan in a shared database.
> > > >  3. I think it is (maybe) posible to 'share' the
> >
> > dialplan through IAX
> >
> > > (anybody ?)
> > >
> > > Your suggestions are reasonable if you know the
> >
> > dialplan. In my case it
> >
> > > can be possible that I will forward a number block
> >
> > to a customer. I have
> >
> > > not (and will not have) any knowledge of the
> >
> > length of the numbers the
> >
> > > customer uses, I only know the base of the block,
> >
> > neither does the
> >
> > > customer have to use an asterisk as termination.
> > >
> > > Example:
> > > I have a block +49-241-9909888 [0-99999]. I
> >
> > forward this block to a
> >
> > > customer. This customer can add one to five digits
> >
> > to this block
> >
> > > depending on his needs and I will never have
> >
> > knowledge of how many
> >
> > > digits he uses.
> > >
> > > As you see, if you want use chan_ss7 as a
> >
> > multi-customer SS7-to-SIP
> >
> > > gateway with a national numbering plan without
> >
> > fixed length numbers (as
> >
> > > in the US) there is no way around a timer. It's
> >
> > sad but true. ;)
> >
> > > >>And last but not least, I also had the problem
> >
> > that no ringback tones
> >
> > > >>were generated by asterisk. The following two
> >
> > lines in the dialplan
> >
> > > >>inserted before the Dial statement do the trick:
> > > >>
> > > >>
> > > >>exten => _X.,n,SetLanguage(de)
> > > >>exten => _X.,n,Playtones(ring)
> > > >
> > > > We actually tried this, but we had to insert a
> >
> > ,1,Answer before the
> >
> > > Playtones command.
> > >
> > > > ...but the Answer before Playtones, breaks most
> >
> > telcos billing system,
> >
> > > > since a call is 'from the Answer to a hangup'.
> > >
> > > It works here without the answer as there is
> >
> > early-Media after receiving
> >
> > > an IAM. This works also with MOH instead of the
> >
> > ringback beeps, what can
> >
> > > be quite funny.
> > >
> > > Best regards,
> > > Kai
> > >
> > > --
> > > Kai Militzer                 WESTEND GmbH  |
> >
> > Internet-Business-Provider
> >
> > > Technik                      CISCO Systems Partner
> >
> > - Authorized Reseller
> >
> > >                               Lütticher Straße 10
> >
> >     Tel 0241/701333-14
> >
> > > km at westend.com               D-52064 Aachen
> >
> >       Fax 0241/911879
> >
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