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@xxxxxxxxx> 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@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 > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by > > > > Easynews.com -- > > > > > asterisk-ss7 mailing list > > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com > > -- > > > > asterisk-ss7 mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection > around http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7
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