The [0-9] Means that the first digit can be any number between 0 and 9. This means ANYTHING will be transferred. What you are doing is creating two dial peers with the same matched digits. What you would really want to do is lets say your DNIS is 2815551200 - 2815551299 you can create a few dial peers which looks like this
dial-peer voice 1 pots description allow DID calling to the Asterisk. If not here, you will get a second dialtone application session incoming called-number [0-9]T direct-inward-dial no register e164 dial-peer voice 2 pots preference 1 application session destination-pattern [0-9]T no digit-strip port 2/0:23 no register e164 dial-peer voice 3 voip destination-pattern 28155512T ( or destination-pattern 28155512..) progress_ind setup enable 3 progress_ind progress enable 8 session protocol sipv2 session target ipv4:xxx.xxx.xxx.xxx dtmf-relay rtp-nte playout-delay minimum low no vad Remember that cisco is different than Asterisk in the handling of calls. Cisco takes the longest matched destination pattern or called-number and attempts to connect with that dial peer first. The [0-9]T is actually only matching 1 digit, the first digit. And the 28155512T is actually matching 8 digits which means if a call came in as 2815551234 it would match more digits to dial peer voice 3. ----------------------- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 [EMAIL PROTECTED] wrote: > The call is inbound on the pots dial-peer, so you should use incoming > called-number, as opposed to destination-pattern. > > dial-peer voice 1 pots > incoming-called number [0-9]T > no digit-strip > direct-inward-dial > port 3/0:D > > I'm not familiar with the [0-9] syntax, but if it works, ok. I > usually use "." > Also, you can specify the sip destination directly in the dial-peer, > which makes using sip with the cisco's more flexible unless you're > using a separate sip proxy. > > session protocol sipv2 > session target ipv4:5.5.5.5 > > -g > > > > > On Wed, 2004-07-14 at 07:27, [EMAIL PROTECTED] wrote: >> >> Hi. >> >> >> If I use a Cisco as a PSTN termination GW and need to route all >> incoming isdn calls to my asterisk and all outgoing calls from >> asterisk via the cisco out to pstn, how do I do that ? >> >> >> in the cisco I have this: >> >> dial-peer voice 1 pots >> destination-pattern [0-9]T >> no digit-strip >> direct-inward-dial >> port 3/0:D >> ! >> dial-peer voice 50 voip >> destination-pattern [0-9] >> voice-class codec 1 >> session protocol sipv2 >> session target sip-server >> no vad >> dtmf-relay rtp-nte >> ! >> >> >> ------- >> >> But theese to dialplans seem to interrupt each other. >> >> When an incoming call from PSTN goes through this the pattern can be >> matched by the first, and then be routed ot on the PSTN again, >> creating a loop. >> >> How do I do this in the smartest and easiest way ? >> >> /Mike >> >> _______________________________________________ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
