I've been scouring the mailing list archives for an answer to this, and cannot find one. I'm hoping someone else out there has run into this.
I'm running a Lucent TNT with TAOS 11.0.2 and trying to get it to process VOIP calls via Asterisk. The TNT is currently accepting dialup calls and functioning normally. My dialup number DIDs are assigned to DNIS profiles and working as expected. I also have a block of DIDs that are not assigned to dialup pools which I intended to use for VOIP calls. Communication between the TNT and Asterisk seems to be operating properly, but I'm unable to accept or originate calls. When I attempt to dial out, I see the following in the TNT's syslog: 10.0.0.10 = TNT 10.0.0.103 = Asterisk Nov 5 14:16:58 tnt0 1/17: SIP Call Admission Control: Incoming new INVITE refused: cause 92.; progress 7.; SIP rsp code 503; Call-Id: [EMAIL PROTECTED]; From-Tag: xxxxxxxxxx Likewise, when I attempt to dial one of my DID's not assigned to a dialup trunk, which I intend to use for VOIP: Nov 5 14:17:33 tnt0 1/17: SIP Call Admission Control: Incoming PSTN Call refused: cause 92.; progress 7.; [MBID 360] My TNT configuration is as follows: new MEDIA-GATEWAY set name = voip set active = yes set protocol-type = sip set mg-sig-address type = specific set mg-sig-address ip-address = 10.0.0.10 set mg-rtp-address type = specific set mg-rtp-address ip-address = 10.0.0.10 set transport-options type = udp set transport-options heartbeat = yes set voip-options codec-options g711-ulaw dtmf-tone-passing = rtp set voip-options codec-options g711-ulaw silence-det-cng = yes set sip-options primary-proxy ip-address = 10.0.0.103 set sip-options primary-proxy transport-options heartbeat = yes set sip-options registration-proxy ip-address = 10.0.0.103 set sip-options unknown-ani = 0000000000 set sip-options unknown-name = Unknown set sip-options blocked-ani = 0000000000 set sip-options blocked-name = Blocked write -f My 12 T1/PRI are configured exactly alike: new T1 set name = PRI-0 set physical-address shelf = shelf-1 set physical-address slot = slot-1 set physical-address item-number = 1 set line-interface enabled = yes set line-interface frame-type = esf set line-interface encoding = b8zs set line-interface signaling-mode = isdn set line-interface default-call-type = dnis-or-voip set line-interface switch-type = nat-isdn-2-pri set line-interface front-end-type = csu set line-interface channel-config 24 channel-usage = d-channel set line-interface collect-incoming-digits = yes set line-interface voip-gain-control output-pad = 9db-loss set line-interface media-gateway = voip set line-interface egress-ani-dnis-format = dnis write -f Asterisk sip.conf: [maxtnt] type=friend host=10.0.0.10 dtmfmode=inband callerid="MaxTNT" <maxtnt> context=toll-access qualify=yes reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw [xxxxxxx] type=friend host=dynamic nat=yes callerid=Name <xxxxxxx> context=toll-access dtmfmode=info call-limit=1 [EMAIL PROTECTED] disallow=all allow=g729 allow=ulaw Asterisk extensions.conf: [toll-trunks] exten => _1NXXNXXXXXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten => _1NXXNXXXXXX,2,Hangup [local-trunks] exten => _NXXXXXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten => _NXXXXXX,2,Hangup [local-access] include => extensions include => local-trunks [toll-access] include => local-access include => toll-trunks I apologize if this is considered off-topic. My thoughts are that I have a problem with the configuration of my TNT and not Asterisk itself. -- +-----------------------------------------------+ ^__^ | Shane DeRidder [EMAIL PROTECTED] | . (oo)\______ | Principal Member http://silicondairy.net/ | o (__)\ )\/\ | Silicon Dairy, LLC. 802.846.4433 x101 | 0 ||---w | +-----------------------------------------------+ || ||
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