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@xxxxxxxxxxx> 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
> L?tticher Stra?e 10 Tel 0241/701333-14
> km@xxxxxxxxxxx D-52064 Aachen Fax 0241/911879
>
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