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

[leonimar cape]

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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