Marc Rys
2005-Dec-08 10:51 UTC
[Asterisk-Users] Lucent MAX TNT - how do I route a DID to my sip trunk
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@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 -- Internal Virus Database is out-of-date. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.5/147 - Release Date: 10/24/2005 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051208/81d779d8/attachment.htm