[Asterisk-Users] Asterisk & Lucent TNT w/11.0.2

[Shane DeRidder]

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:
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx at 10.0.0.103; 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
mailbox=xxxxxxx at default
disallow=all
allow=g729
allow=ulaw

Asterisk extensions.conf:

[toll-trunks]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@10.0.0.10,60)
exten => _1NXXNXXXXXX,2,Hangup

[local-trunks]
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@10.0.0.10,60)
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       shane at silicondairy.net   |   . (oo)\______
| Principal Member     http://silicondairy.net/ |  o  (__)\      )\/\
| Silicon Dairy, LLC.  802.846.4433 x101        | 0        ||---w |
+-----------------------------------------------+          ||    ||
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