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 <[EMAIL PROTECTED]> 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
[EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879
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