Probably by preference and peer type matching, try setting a new VoIP peer for inbound calls from asterisk
LTenorio > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Ivan Vershigora > Sent: Thursday, November 03, 2005 10:27 AM > To: [email protected] > Subject: [Asterisk-Users] call from asterisk to SIP cisco 5300 > > > i dial on my phone to to 80912222222 > and convert it on asterisk to #00#70912222222 But Cisco says 404 > > ============cisco peer============= > ! > dial-peer voice 22 pots > huntstop > preference 5 > destination-pattern #00#......\* > translate-outgoing calling 1 > direct-inward-dial > port 0:D > prefix 810 > ! > ================================ > > ============peer in sip.conf========== > [krdvox] > context=from-sip > type=peer > host=123.123.123.123 > canreinvite=yes > dtmfmode=inband > ================================ > > ============extensions.conf========== > exten => _.,1,SetCallerID("8612730000" <8612731107>[|a]) > exten => _.,2,Dial(SIP/#00#7${EXTEN:[EMAIL PROTECTED],60) > exten => _.,3,Congestion > ================================ > > ============Asterisk says=========== > -- Executing Dial("SIP/201-2966", > "SIP/[EMAIL PROTECTED]|60") in new stack > -- Called [EMAIL PROTECTED] > -- Got SIP response 404 "Not Found" back from XXX.XXX.XXX.XXX > -- SIP/krdvox-3910 is circuit-busy > == Everyone is busy/congested at this time > =============================== > > ======CISCO debug ccsip =========== > Nov 3 16:10:03.516: Received: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34 > From: "8612730000" <sip:[EMAIL PROTECTED]>;tag=as74db268c > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: CSCO/6 > Date: Thu, 03 Nov 2005 13:10:06 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 235 > > ..... > > Nov 3 16:10:03.524: MatchNextPeer: Peer 999 matched Nov 3 > 16:10:03.524: Using Voice Class Codec, tag=1 > > ..... > > Disconnect Cause (SIP) : 404 > > =============================== > Nov 3 16:10:03.524: MatchNextPeer: Peer 999 matched > > Peer 999- wrong one !!!!!!! > why he cant find dial-peer voice 22 > > > ???????????????????? > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
