[Asterisk-Users] Lucent MAX TNT - how do I route a DID to my sip trunk

Thu Dec 8 10:51:10 MST 2005

Currently I’m running asterisk @ home 1.5 and a Lucent Max TNT.  I want to use the Max as a PSTN gateway for @home.  To do this I have a PRI terminated to the Max TNT.

As you can see below I have established a SIP trunk between @home and the MAX TNT.

asterisk1*CLI> sip show peers

Name/username    Host            Dyn Nat ACL Mask             Port     Status

maxtrunk1        172.16.255.191              255.255.255.255  5060     OK (15 ms)

230/230          172.16.255.200   D   N      255.255.255.255  20924    Unmonitored

200/200          (Unspecified)    D          255.255.255.255  0        Unmonitored

asterisk1*CLI>

>From my softphone (ext. 230) I can dial out the Max TNT successfully.  I have setup a DID pointing to my softphone extension.  E.G.   NPA-NXX-0230 -> ext. 230.

Of course the DID terminates on the PRI connected to the Max TNT.  But when I call NPA-NXX-0230 from an outside PSTN line, I get this message on the MAX.

LOG info, Shelf 1, Controller, Time: 14:40:28--

  Releasing <1f12e4c2-39-1df9a85c at 172.16.255.191>: Calling = NPANXX3405,Called =

 NPANXX0230, Q850 Cause = 1,Sip Response = 404 (Not Found),Progress Cause = NONE

LOG warning, Shelf 1, Slot 3, Time: 14:40:28--

  [1/3/67/0] STOP: ''; cause 801.; progress 1404.; host 0.0.0.0 [MBID 71; NPANXX

3405->NPANXX0230]

I don’t see any debug information come across my terminal session with @home when I attempt to make the call.

What is necessary to make the Max TNT route the call to @home when receiving a call for NPA-NXX-0230? And what do I need to do to route 100 DID’s to my @home box?  Where in the Max do I put the range of DID’s allocated to me and have the calls destined for them get passed onto my @home box?  Any help is greatly appreciated.

Marc

Below is most of the meat of my Max TNT’s config.,,,,

[in MEDIA-GATEWAY/voip]

name* = voip

active = yes

protocol-type = sip

mgc-address = [ { "" 0.0.0.0 2944 } { "" 0.0.0.0 2944 } { "" 0.0.0.0 2944 } { "+

mg-sig-address = { interface-dependent 0.0.0.0 }

mg-rtp-address = { system-default 0.0.0.0 }

h248-options = { text 3000 { no 0 } { 8000 6000 9000 [ { "" "" } { "" "" } { ""+

ipdc-options = { "" IASCTNT1B { sig-queue-depth 60 send-info-to-mgc 120 reject-+

transport-options = { udp no { 0 1000 3000 30000 7 6 } }

voip-options = { g711-ulaw { { yes 4 rtp yes } { yes 4 inband no } { no 1 rtp n+

dialed-gw-options = { disabled disabled disabled yes ring-tone-on-alerting disa+

rt-fax-options = { no yes yes yes yes 0 no 14400 no }

tos-rtp-options = { no precedence-tos 00 000 normal }

tos-sig-options = { no precedence-tos 00 000 normal }

sip-options = { 500 4000 6 10 60 { 172.16.255.87 "" 5060 compact { udp no { 0 0+

call-admission-control-options = { { yes } }

[in MEDIA-GATEWAY/voip:sip-options]

t1-timer = 500

t2-timer = 4000

invite-retries = 6

non-invite-retries = 10

tcp-idle-timer = 60

primary-proxy = { 172.16.255.87 "" 5060 compact { udp no { 0 0 0 0 0 0 } } }

secondary-proxy = { 0.0.0.0 "" 5060 compact { udp no { 0 0 0 0 0 0 } } }

registration-proxy = { 172.16.255.87 "" 5060 compact { udp no { 0 0 0 0 0 0 } }+

proxy-heartbeat = 0

proxy-failover-window = 60

reroute-on-proxy-failure = no

trusted-proxy = { disabled [ { "" 0.0.0.0 } { "" 0.0.0.0 } { "" 0.0.0.0 } { "" +

unknown-ani = 0000000000

unknown-name = www.rystec.com

blocked-ani = 0000000000

blocked-name = blocked

privacy-proxy-require = disabled

isdn2sip-mapping = [ { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 } { +

sip2isdn-mapping = [ { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 } { 0 0 } { +

start-call-method = invite

trunk-group-options = { prepend-to-userinfo "" no prepend-to-userinfo "" }

onhold-minutes = 0

support-100rel = disabled

internationalize = no

international-prefix = no

country-code = ""

national-destination-code = ""

local-number-ton = unknown-ton

notify-timer = 0

options-trigger = [ { 488 304 } { 488 305 } { 606 304 } { 606 305 } { 415 304 }+

invite-with-multiple-codecs = disabled

egress-call-duration = 0

magic-number-prefix = ""

send-optional-headers = yes

user-agent-info = Lucent-Universal-Gateway

server-info = Lucent-Universal-Gateway

internationalize-cas = yes

T1/{ shelf-1 slot-2 1 } read

admin> list

[in T1/{ shelf-1 slot-2 1 }]

name = ASTERISK-PRI-01

physical-address* = { shelf-1 slot-2 1 }

line-interface = { yes esf b8zs eligible middle-priority isdn te wink-start dni+

autogenerated = no

[in T1/{ shelf-1 slot-2 1 }:line-interface]

enabled = yes

frame-type = esf

encoding = b8zs

clock-source = eligible

clock-priority = middle-priority

signaling-mode = isdn

isdn-emulation-side = te

robbed-bit-mode = wink-start

default-call-type = voip

switch-type = att-pri

nfas-group-id = 0

nfas-id = 0

incoming-call-handling = internal-processing

call-by-call = 0

network-specific-facilities = 0

data-sense = normal

idle-mode = flag-idle

FDL = none

front-end-type = dsx

DSX-line-length = 1-133

CSU-build-out = 0-db

overlap-receiving = no

pri-prefix-number = ""

tx-clir-flag-in-voip = no

trailing-digits = 2

t302-timer = 10000

channel-config = [ { switched-channel 9 "" 1 255 } { switched-channel 9 "" 1 25+

maintenance-state = no

input-sample-count = one-sample

sendDisc-val = 0

hunt-grp-phone-number-1 = ""

hunt-grp-phone-number-2 = ""

hunt-grp-phone-number-3 = ""

collect-incoming-digits = no

t1-inter-digit-timeout = 3000

r1-use-anir = no

r1-first-digit-timer = 340

r1-anir-delay = 350

r1-anir-timer = 200

r1-modified = no

first-ds0 = 0

last-ds0 = 0

nailed-group = 32768

ss7-continuity = { loopback single-tone-2010 }

down-trans-delay = 25

up-trans-delay = 100

t200-timer = 2000

t203-timer = 30000

voip-gain-control = { 0db 0db }

media-gateway = voip

status-change-trap-enable = no

cause-code-verification-enable = yes

g711-voice-natural = no

use-ds1-idle-pattern = no

idle-pattern = 255

two-b-channel-transfer-options = never-use-tbct

egress-ani-dnis-format = dnis

send-dnis-type-of-number = national

send-dnis-numbering-plan = isdn-telephony

isdn-calling-name-delivery = off

media-on-disconnect-progress = yes

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