Hi<br><br>I fully support the Timeout parameter as this is a common practice in SIP based communication.<br><br>I work a lot with Patton SmartNode Sip Gateways and in the configuration we have the following.<br><br>context cs switch
<br> digit-collection timeout 3<br> routing-table called-e164 TEST1<br> route .T dest-interface IF_E1<br> route 00.% dest-interface IF_E2<br><br>On many SIP phones you also have the option to choose Timeout or <i>
Early Dial </i>(484 response)<br><br>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.<br><br>-- <br>Are Casilla<br><a href="http://astartelecom.com">
http://astartelecom.com</a> - Independent VOIP Telecoms Broker. Asterisk Consultants<br><a href="http://astbill.com">http://astbill.com</a> - Open Source Billing, Routing and Management software for Asterisk and VOIP<br>AstBill DEMO:
<a href="http://demo.astbill.com">http://demo.astbill.com</a><br><br><div><span class="gmail_quote">On 3/16/06, <b class="gmail_sendername">Kai Militzer</b> <<a href="mailto:km@westend.com">km@westend.com</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello Jacob, hello all,<br><br>Jacob Tinning wrote:<br><br>> We didn't like the timer-solution because we think its wrong to delay all calls
<br>> X seconds just because the SS7-asterisk doesn't know another Asterisk's dialplan.<br><br>Thats why I made it configurable, so that it can be turned off, when not<br>needed. ;)<br><br>> My suggestions is<br>> 1. Use identical dialplans on the SS7-gateway and the SIP server
<br>> 2. Store the dialplan in a shared database.<br>> 3. I think it is (maybe) posible to 'share' the dialplan through IAX (anybody ?)<br><br>Your suggestions are reasonable if you know the dialplan. In my case it
<br>can be possible that I will forward a number block to a customer. I have<br>not (and will not have) any knowledge of the length of the numbers the<br>customer uses, I only know the base of the block, neither does the<br>
customer have to use an asterisk as termination.<br><br>Example:<br>I have a block +49-241-9909888 [0-99999]. I forward this block to a<br>customer. This customer can add one to five digits to this block<br>depending on his needs and I will never have knowledge of how many
<br>digits he uses.<br><br>As you see, if you want use chan_ss7 as a multi-customer SS7-to-SIP<br>gateway with a national numbering plan without fixed length numbers (as<br>in the US) there is no way around a timer. It's sad but true. ;)
<br><br>>>And last but not least, I also had the problem that no ringback tones<br>>>were generated by asterisk. The following two lines in the dialplan<br>>>inserted before the Dial statement do the trick:
<br>><br>><br>>>exten => _X.,n,SetLanguage(de)<br>>>exten => _X.,n,Playtones(ring)<br>><br>><br>> We actually tried this, but we had to insert a ,1,Answer before the Playtones command.<br>
> ...but the Answer before Playtones, breaks most telcos billing system,<br>> since a call is 'from the Answer to a hangup'.<br><br>It works here without the answer as there is early-Media after receiving<br>an IAM. This works also with MOH instead of the ringback beeps, what can
<br>be quite funny.<br><br>Best regards,<br>Kai<br><br>--<br>Kai Militzer WESTEND GmbH | Internet-Business-Provider<br>Technik CISCO Systems Partner - Authorized Reseller<br> Lütticher Straße 10 Tel 0241/701333-14
<br><a href="mailto:km@westend.com">km@westend.com</a> D-52064 Aachen Fax 0241/911879<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
Easynews.com</a> --<br><br>asterisk-ss7 mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a>
<br></blockquote></div><br><br clear="all"><br><br>