OpenScape 4000 - V7 UC
HiPath 4000 - V6
Unified Communications Linux vServer (OS4k)
Linux Communication Server (HP4k)
Native SIP
Trunk - VoIP Guide
Contents: 1. Scope 2. Card (Board) Configuration: Configuration: 2.1.SUITABLE CARDS 2.2. 2.2. AMO-BFDAT = Block Functional DATa for STMI Cards 2.3. 2.3. AMO-BCSU = Board/Card Config. [ SU=Switching Unit ] 2.4. AMO-CGWB AMO-CGWB = Configuration of GateWay STMI Board/card; options: 2.5. 2.5. INSERT the STMI card with Lan cable. 3. TRUNK CONFIGURATION CONFIGURATION - TDM/RMX TDM/RMX SIDE.... 3.1. 3.1. AMO-COSSU = Class Of Service for the Trunk 3.2. AMO-COT = Class of Trunk for Call Processing 3.3. AMO-COP = Class of Parameter for Device Handler. 3.4. 3.4. AMO-BUEND = Buendel = Bundle = T runk Group 3.5. 3.5. AMO-TDCSU = Trunk Digital Config. 4. LCR ROUTING ( no gkreg 4.1 /2/3 ) : 4.1. 4.1. Open Number Routing: LCR with TIE Access Code + hear "external" Dial Tone (W) + SIP Provider number (Z) + # 4.2. 4.2. Closed Number Routing: Dial Sip Vendor Devices (stn. number 7170 to 7199) directly... 4.3. 4.3. One Number Routing Access: LCR outbound (outgoing) Sip with one fixed Number to dial. 4.4. Routing using GKREG; one STMI multiple IP destinations; no sip profile used. 4.5. 4.5. Miscelaneous AMO's for US config. 4.6. Ways to convert incoming DID numbers. 5. SIP GW Web Browser Management Browser Management (WBM) Settings: Settings: 5.1. 5.1. Activation of SIP Trk Profile GW. 5.2. 5.2. Other possible SIP GW settings 6. Troubleshooting Troubleshooting. 6.1. Wireshark 6.2. 6.2. RTDS, internal TDCSU ISDN trace 6.3. 6.3. Maintenance AMO 6.4. 6.4. RMX - Traces 6.5. 6.5. STMI - GW Traces 6.6. STMI STMI LoadWare ( LW ) 6.7. STMI - CLI = Command Line Interface 7. SIP (STMI) x ISDN I SDN (TDCSU) messages conversion.
Changes: 1.Published = July 07, 2014. 2.Rev.=July = added item 4.4. Routing with GKREG, GKREG , one STMI multiple IP destinations and item 6.6 LW. 3.Rev.=Sept.=m atch channels cgwb=buend=tdcs u=e.g.12 , all chann. busy = 503 Service Not Available & minor changes. 4.Update=March 2015 = added cgw comments about ports & RFC 5.Upd.=June 2015 = NatTrkEnterprise best profile for non certified apps + 1 STMI multiple GKREG destinations is supported + item 4.6 Repository:
KM & http://4000.g lobal-intra.net
Total of 26 pages.
[email protected] Tec.Support Eng. @ Global Operations (G0) Irving - TX 75038 USA
1
1. Scope: Guide for Installation and Maintenance of NATIVE SIP TRUNKS ; VoIP (Voice over Internet Protocol) Stn. --> OS4k --> STMI GW card [ tdm/isdn trk <= conv. conv. => => SIP ] ==> Lan.Netw. --> Cloud --> VoIP Provider 2. Card (Board) Configuration: 2.1. SUITABLE IP GW CARDS
Q2316-X Q2316-X10 Q2324-X500 Q2324-X510 Q2324-X511
186 Q2330-X
IPGW IPGW IPGW IPGW IPGW
STMI2 STMI2 STMI4 STMI4 STMI4
1 1 1 1 1
60-BChannels 120 120 60 60 120 120 120
LG98/PZKSTI40 => /* LW are stored on :PDS:APSP/LTG/LGxx/PZKyyyxx :PDS:APSP/LTG/LGxx/PZKyyyxx /* LG98/PZKSTI40 LG98/PZKSTI40 LG98/PZKSTI40 LG98/PZKSTI40
IPGW
vHG3500
1
120
LGA0/PZKSGVB0
=> SGVB=SoftGate Virtual Board
2.2. AMO-BFDAT AMO-BFDAT = Block Functional Functional DATa for STMI Cards.
ADD-BFDAT:FCTBLK=11,FUNCTION=HG3550&HG3530&SIP,BRDBCHL=BCHL120; CHANGE-BFDAT:CONFIG=CONT,FCTBLK=11,FUNCTION=HG3550,LINECNT=2,UNITS=3; /* LineCouNT=PEN /*set 2 lines=circ.(max.4), e.g. pen=1-3-5-0 & -1; each PEN 3x(unit=10)= 30BChan.(max.120BChan.)
CHANGE-BFDAT:CONFIG=CONT,FCTBLK=11,FUNCTION=HG3530,LINECNT=51,BCHLCNT=45; /* above the HFA IP Phones configuration choosed additionaly to the IP Trk../* /*on this example can add 51 HFA ports; however maximal 45 stn. can talk simultaneously /* CHANGE-BFDAT:CONFIG=CONT,FCTBLK=11,FUNCTION=SIP,LINECNT=8,BCHLCNT=6; /* SIP Phones CHANGE-BFDAT:CONFIG=OK,FCTBLK=11,ANSW=YES; /*OK, closing no more changes possible... -If STMI card has only 60 BChan. choose BRDBCHL=BCHL60; -If you choose BRDBCHL=BCHL60&BCHL120; you can use STMI with 60 or 120 Channels; however you limit the channels to 60 no matter the card you're going to use...
-If card is exclusive for IP Trk (max.120BChann.), BFDAT does not need the HG3530 & SIP entries.. -For SoftGate vSTMI, you can not mix functions. ADD-BFDAT:FCTBLK=12,FUNCTI ADD-BFDAT:FCTBLK=12,FUNCTION=HG3550,BRD ON=HG3550,BRDBCHL=BCHL120, BCHL=BCHL120,ATTR="SOCO"; ATTR="SOCO";
2 lines=pen=circ. x 30 BChann.= 60 Total | 2. FUNCT : HG3530 51 LINES UNITS BCHLCNT 45 TOTAL BCHAN 45 | | 3. FUNCT : SIP 8 LINES UNITS BCHLCNT 6 TOTAL BCHAN 6 | ----------------------------------------------------------------------------------------------------------------------------------------------------------
2.3. AMO-BCSU = BOARD/CARD CONFIGURATION CONFIGURATION SWITCHING UNIT
ADD-BCSU:MTYPE=IPGW,LTG=1,LTU=1,SLOT=11,PARTNO="Q2316-X10",FCTBLK=11; H34: DUE TO ACTIVATED FEATURE SIGNALLING & PAYLOAD ENCRYPTION AND/OR ACTIVATED DMC THE NUMBER OF CONFIGURED B-CHANNELS FOR HG3530 HAVE BEEN REDUCED BY 11 B-CHANNELS. ACTUAL AVAILABLE HG3530 B-CHANNELS: 34 H34: DUE TO ACTIVATED FEATURE SIGNALLING & PAYLOAD ENCRYPTION AND/OR ACTIVATED DMC THE NUMBER OF CONFIGURED B-CHANNELS FOR SIP HAVE BEEN REDUCED BY 2 B-CHANNELS. ACTUAL AVAILABLE SIP B-CHANNELS: 4 H34: DUE TO ACTIVATED FEATURE SIGNALLING & PAYLOAD ENCRYPTION AND/OR ACTIVATED DMC THE NUMBER OF CONFIGURED B-CHANNELS FOR HG3570 HAVE BEEN REDUCED BY 2 B-CHANNELS. ACTUAL AVAILABLE HG3570 B-CHANNELS: 0 card +-----------------------+--------------------------------+-+---------+-+------------+-----------+------------+-----------+--------------------| IP ADDRESS : 0. 0. 0. 0 B-CHANNELS : 111 BCHLCNT : 98 | IP MODE : IPV4 DHCP V4 : NO DHCP V6 : NO | BLOCK NO : 11 PRERESERVED LINES ASSIGNED : NO | 1. FUNCT : HG3550 2 LINES B-CHANNELS : 60 BCHLCNT : 60 | 2. FUNCT : HG3530 51 LINES B-CHANNELS : 45 BCHLCNT : 34 | 3. FUNCT : SIP 8 LINES B-CHANNELS : 6 BCHLCNT : 4 +-----------------------+--------------------------------+-+---------+-+------------+-----------+------------+-----------+---------------------
Not PResent
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2.4. AMO-CGWB = Configuration of GateWay STMI Board/card; options:
CHANGE-ZANDE:TYPE=ALLDATA,GATEKPR=YES; /* Internal Gate Keeper for the STMI cards. /* ADD-CGWB:LTU=1,SLOT=11,SMODE=NORMAL,IPADR=172.19.208.184,NETMASK=255.255.255.128, PATTERN=255,DEFRT=172.19.208.129,BITRATE="100MBFD",TRPRSIP=12; /* Had 12 SIP Trk channels/call path available at SIP Service Provider /*
CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,T38FAX=YES,RFCFMOIP=YES, RFCDTMF=YES,REDRFCTN=YES,PRIO=PRIO1,CODEC=G711U,RTP="20"; CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,PRIO=PRIO2,CODEC=G711A,RTP="20"; CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,PRIO=PRIO3,CODEC=G723,RTP="30"; CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,PRIO=PRIO4,CODEC=G729,RTP="20"; CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,PRIO=PRIO5,CODEC=G729A,RTP="20"; CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,PRIO=PRIO6,CODEC=G729B,VAD=YES,RTP="20"; CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=ASC,PRIO=PRIO7,CODEC=G729AB,VAD=YES,RTP="20"; /* set all possible codec types, that stmi gw can support thru..../* /* The talk codec will be negotiated and choosed by end devices /*
CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=MGNTDATA,MGNTIP=172.19.208.140, BUSIP=172.19.208.140; /* 172.19.208.140 = Assistant IP to upload/get STMI CARD SETTINGS from backup; /* WHEN load/replace the CARD..../* TCP signal. Port (default=4060) NETMASK = 255.255.255.128 VLAN = NO (NO) DEFRT = 172.19 .208.129 (0.0.0.0 = NOT CONFIGURED) BITRATE = 100MBFD (AUTONEG ) VLANID = 0 (0) PATTERN = 255 (213) TLSP = 4061 (4061) => 255 = uLaw for USA , 213= A-Law TRPRSIP = 12 (0) TRPRH323 = 0 (0) => TRPRSIP=define TRunk PRotocol SIP, 12Chann. config. (max.120) . TRPRSIPQ = 0 (0) TPRH323A = 0 (0) => 12 = SIP Provider call path/Channels/ports available ... DNSIPADR2 = 0 .0 .0 .0 SIPTLSP = 5061 (5061) Can set only 1 trk prot.per STMI: SIP or SIPQ or H323(XPress.) USEWANIF = NO (NO) ASC DATA - CONFIGURABLE VALUES: => Audio Stream Control ------------------------------TOSPL = 184 (184) TOSSIGNL = 104 (104) UDPPRTLO = 29100 (29100) UDPPRTHI = 30099 (30099) => UDP Voice payload Ports Low (std.=29100) + 1k = High (30099) T38FAX = YES (YES) REDRFCTN = YES (YES) => ask provider if they comply RFC 2198 = Redundant of rfc2833 dtmf tones. RFCFMOIP = YES (NO) RFCDTMF = YES (YES) => RFC2833 Fax Modem and DTMF tones (RFC = Request For Comments = "norm") PRIO1 PRIO2 PRIO3 PRIO4 PRIO5 PRIO6 PRIO7
: : : : : : :
CODEC CODEC CODEC CODEC CODEC CODEC CODEC
= = = = = = =
G711U G711A G723 G729 G729A G729B G729AB
VAD VAD VAD VAD VAD VAD VAD
= = = = = = =
NO NO NO NO NO YES YES
RTP-SIZE RTP-SIZE RTP-SIZE RTP-SIZE RTP-SIZE RTP-SIZE RTP-SIZE
MANAGEMENT STATION AND BACK-UP SERVER ------------------------------------MGNTIP = 172.19.208.140 (0.0.0.0) BUSIP = 172.19.208. 140 (0.0.0.0)
= = = = = = =
20 20 30 20 20 20 20
=> G711u-Law= USA/Canada/Japan = PCM24 = TDM Quantization = high Bandwidth => G711A = All other countries (Germany, Europe, Latin America..)= PCM30 => G723 = barely used
=> Assistant IP address for backup/restore
2.5. INSERT the STMI card with Lan cable. F5350 E8 N9300 IN SERV BPA BOARD SECONDARY IN ATTEND L ALARM CLASS:CENTRAL:002 ** :LTG1 :LTU1 :011: 0 : 0 Q2316-X10 STMI2 F5749 E8 N9301 NO ACT BPA BOARD ALARM CLASS:CENTRAL:002 REASON:00H ONLY SIGNALING MSG_HBR_WARNING
14-06-13 11:00:04 BST:01
PLS:-08
LW REQUEST
14-06-13 11:01:15
F5749 E8 N9303 NO ACT BPA BOARD LW REQUEST ALARM CLASS:CENTRAL:002 ** :LTG1 :LTU1 :011: 0 : 0 Q2316-X10 REASON:00H ONLY SIGNALING MSG_IP_LINK_RESTORE
14-06-13 11:01:16 STMI2/ 1
F5471 E8 N9304 IN SERV BPA CIRCUIT L1 PORT ACTIVE ALARM CLASS:CENTRAL:002 ** :LTG1 :LTU1 :011: 0- 60: 0 Q2316-X10 STMI2/1 REASON:07H LAYER 1 ACTIVE
BST:01
PLS:-08
14-06-13 11:01:17 BST:01
PLS:-08
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3. TRUNK CONFIGURATION - RMX-TDM = ISDN inSIDE STMI-GW = Switching Unit = SU = SWU.... 3.1. AMO-COSSU = Class Of Service Switching Unit for the Trunk
ADD-COSSU:NEWCOS=11,INFO="Native SIP - Unify"; CHANGE-COSSU:TYPE=COS,COS=11,AVCE=TA&TNOTCR&MB&FWDNWK&TTT&FWDECA; CHANGE-COSSU:TYPE=COS,COS=11,AFAX=TA&TNOTCR&BASIC; CHANGE-COSSU:TYPE=COS,COS=11,ADTE=TA&TNOTCR&BASIC; NATIVE SIP - UNIFY | | | TA | TA | TA | | | TNOTCR | TNOTCR | TNOTCR | | | MB | BASIC | BASIC | | | FWDNWK | | | | | TTT | | | | | FWDECA | | | +------+-----------------+---------------+-----------------+
3.2. AMO-COT = Class of Trunk for Call Processing
ADD-COT:COTNO=11,PAR=RCL&XFER&ANS&CHRT&CEBC&BSHT&BLOC&LWNC&NLCR&TSCS&DFNN&NLRD& LINC&NOFT&IBBA&NTON; CHANGE-COT:COTNO=11,COTTYPE=COTADD,DEV=INDEP,INFO="Native SIP - Unify";
RCL XFER ANS CHRT CEBC BSHT BLOC LWNC NLCR TSCS DFNN NLRD LINC NOFT IBBA NTON
3.3. AMO-COP = Class of Parameter for Device Handler.
ADD-COP:COPNO=11,PAR=L3AR,TRK=TA,TOLL=TA; CHANGE-COP:COPNO=11,COPTYPE=COPADD,DEV=INDEP,INFO=" Native SIP - Unify";
L3AR
CO TRUNK ACCESS: TRUNK ACCESS
TA
TOLL ACCESS: TRUNK ACCESS
TA
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3.4. AMO-BUEND =
Buendel = Bundle =
Trunk Group
ADD-BUEND:TGRP=11,NAME="NATIV.SIP_UNIFY",NO=12; /* NO=12 =>how many ports/licenses/channels/call path available at Sip Trk Provider or SIP external Device (e.g.Fax Server).
3.5. AMO-TDCSU =
Trunk Digital Config. Switching Unit
ADD-TDCSU:OPT=NEW,PEN=1-1-11-0,COTNO=11,COPNO=11,LCOSV=1,LCOSD=1,COS=11,CCT=" NATIVE SIP",PROTVAR="ECMAV2",SEGMENT=8,NNO=11 ,TGRP=11,DEV=HG3550IP,SRCHMODE=CIR, BCHAN=1&&12,BCGR=1; /* CHAN=channels available at SIP Provider, e.g.: 12 ) H04: AT PEN 1-01-011-0 ASSIGNED TO BCGR 1:
THE FOLLOWING B-CHANNELS HAVE BEEN NEWLY 1 && 12
F5418 E8 N9782 IN SERV BPB CIRCUIT LX ACTIVE ALARM CLASS:SWU-PER:001 ** :LTG1 :LTU1 :011: 0 : 0 Q2316-X10 REASON:07H LAYER 1 ACTIVE (GREEN) FORMAT:36 DEVICE NAME: HG3550IP
14-06-13 15:44:59 STMI2/1
F5418 E8 N9783 IN SERV BPB CIRCUIT LX ACTIVE ALARM CLASS:SWU-PER:001 ** :LTG1 :LTU1 :011: 0 : 0 Q2316-X10 STMI2/1 REASON:09H LAYER 3 ACTIVE FORMAT:36 DEVICE NAME: HG3550IP
BST:01
PLS:-08
14-06-13 15:44:59 BST:01
PLS:-08
if 0 zero, | SEGMENT = 8 DEDSCC = DEDSVC = NONE | | FACILITY = DITIDX = SRTIDX = | | TRTBL SIDANI = N ATNTYP = TIE | = GDTR | CBMATTR = NONE NWMUXTIM = 10 TCHARG = N | | SUPPRESS = 0 DGTPR = CHIMAP = N | | ISDNIP = ISDNNP = | | PNPL2P = PNPL1P = PNPAC = | | TRACOUNT = 31 SATCOUNT = MANY NNO = 11 | | ALARMNO = 0 FIDX = 1 CARRIER = 1 | | ZONE = EMPTY COTX = 11 FWDX = 1 | | DOMTYPE = DOMAINNO = TPROFNO = | | INIGHT = CCHDL = | | UUSCCX = 16 UUSCCY = 8 FNIDX = 1 | | CLASSMRK = EC & G711 & G729AOPT SRCGRP = ( ) | | TCCID = SECLEVEL = SECURE | | HMUSIC = 0 CALLTIM = 60 WARNTIM = 60 | |---------------------------------------------------------------------------| | BCNEG = N BCGR = 1 LWPAR = | | LWPP = 0 LWLT = 0 LWPS = 0 | | LWR1 = 0 LWR2 = 0 | | DMCALLWD = N VNNO = | | SVCDOM = | | BCHAN => 12 available call path at SIP Provider | = 1 && 12 +---------------------------------------------------------------------------+ AMOUNT OF B-CHANNELS IN THIS DISPLAY-OUTPUT: 12
=> PROTVAR <=== copy-PRODE:pvcd,16; for USA
trk loopback is possible, but no loop detection (traccount)
4. Routing. AMO-WABE = WAhl BEwertung = Dialed Digit Translation, No -> Dial PLan Number = stn. or feature AMO-RICHT = Richtung = Direction = Route number, Buend/Trk Group, Destination Number (No). AMO-LODR = LCR (Least Coust Routing) OutDial Rules. AMO-LDAT = Lcr DATa = richt + lodr + attributes + cot index number for outgoing calls. amo-lprof = lcr profiles for ipda/source group (not used here for simplicity) AMO-LDPLN = Lcr Dial PLaN = wabe -> outgoing lcr routing. !! Since V4 Native Sip Trunk is recommended to use Profiles ( only one IP destination ) due to security reasons ; the usage of amo-GKREG IP destinations addr. used on V3 is and still supported by GVS; see details on item 4.4.Routing with GKREG !!!
GKREG:number, ip destin.; + LDAT: destin. GW1=GKREG_number; same as used for SIPQ .....
5
4.1. Open Number Routing: LCR with TIE Access Code + hear "external" Dial Tone (W) + SIP Provider number (Z) + # e.g.: Dial 82 + SIP vendor stn. number + # (end of dial).
ADD-RICHT:MODE=LRTENEW,LRTE=82,LSVC=VCE,NAME="NATIVE SIP ",TGRP=11,DNNO=82, DTMFCNV=FIX,DTMFPULS=PP300,DESTNO=82,PDNNO=82;
ADD-LODR:ODR=12,CMD=ECHO,FIELD=3; ADD-LODR:ODR=12,CMD=END; ADD-LODR:ODR=12,INFO="SIP TRK ACC.CODE";
=> LDPLN 82-W-Z =ECHO field 3 = ECHO Z = Send out only Z => Field 1= 82 Tie; Field 2= W; Field 3 = Z = any number/quan tity
ADD-LDAT:LROUTE=82,LSVC=VCE,LVAL=1,TGRP=11,ODR=12,LAUTH=1,LATTR=WCHREG&PUBNUM, COTIDX=11; With CHarge REGistration (CDR) | | | | | PUBNUM | => PUBlic NUMber = calling #, from SDAT (e.g.: 972.756.0123). +------------------------------------------------------------------------------+
ADD-WABE:CD=82,DAR=TIE;
ADD-LDPLN:LCRCONF=LCRPATT,DIPLNUM=0,LDP="82"-"W"-"Z",DPLN=0,LROUTE=82,LAUTH=1; | +--------+-------------------------------------+------------+ | LDPNO | LDP | DIPLNUM | +--------+-------------------------------------+------------+ | 41 | 82-W-Z | 0 | +-------+--------------------------------------------------+
/* If SIP vendor has fixed numbers, e.g. 7170 to 7199, you can replace the Z by XXXX (4X not 2X)...
6
4.2. Closed Number Routing ( Network LCR ) : LCR outbound (outgoing) Native Sip with # Closed Number Plan # Dial Sip Vendor Devices (e.g. numbers 7170 to 7199) directly...
ADD-WABE:CD=83,DAR=NETRTE,CHECK=N;
ADD-RICHT:MODE=CD,LRTE=83,CD=83,CPS=0,SVC=vce,NAME="N.SIP CLOS# DESTNO=83,DNNO=83,PDNNO=83;
",TGRP1=11,
ADD-LODR:ODR=83,CMD=ECHOALL; ADD-LODR:ODR=83,CMD=END; ADD-LODR:ODR=83,INFO="NATIVE SIP CLOSED #";
ADD-LDAT:LROUTE=83,LSVC=vce,LVAL=1,TGRP=11,ODR=83,LAUTH=1,LATTR=WCHREG,COTIDX=11;
ADD-WABE:CD=7170&&7199,DAR=Stn; /* if other side change/add #s, must done also here /* CHANGE-WABE:CD=7170&&7199,DESTNO=83;
/* The routing 4.1 Open # and 4.2 Closed # and 4.3 One # can be configured together; works as separated routes.../*
7
4.3. One Number Routing Access: LCR outbound (outgoing) Sip with one fixed Number e.g.: dial 811 to connect. ADD-RICHT:MODE=LRTENEW,LRTE=811,LSVC=VCE,NAME="XP AUDIO CONF. ",TGRP=11,DNNO=11, DTMFCNV=FIX,DTMFPULS=PP300,DESTNO=11,PDNNO=11,COTIDX=11;
ADD-LODR:ODR=90,CMD=ECHO,FIELD=1; ADD-LODR:ODR=90,CMD=NPI,NPI=UNKNOWN,TON=UNKNOWN; ADD-LODR:ODR=90,CMD=END; ADD-LODR:ODR=90,INFO="XPRESSION IP";
ADD-LDAT:LROUTE=811,LSVC=VCE,LVAL=1,TGRP=11,ODR=90,LAUTH=1,LATTR=PUBNUM,DNNO=11;
ADD-WABE:CD=811,DAR=TIE,CHECK=N;
ADD-LDPLN:LCRCONF=LCRPATT,DIPLNUM=0,LDP="811",LROUTE=811,LAUTH=1;
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4.4. Routing with GKREG; one STMI multiple IP destinations. - Customers migrating from early versions using only one STMI with multiple sIP destinations can still use this GKREG feature; - This is and still fully supported by Development/GVS ; they recommend using profiles for security reasons i.e. without them, the SIP board can affectively be addressed from any IP. The SIP Profile usage had the disavantage of allow only one SIP far end destination.
4.4.1. Additional amo changes: CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=LEGKDATA,GWNO=11,REGEXTGK=NO; ADD-GKREG:GWNO=11,GWATTR=INTGW&HG3550V2&SIP,DIPLNUM=0,DPLN=0,LAUTH=1 INFO="STMI LTU1 SLOT11 INTERNAL REG. NAT.SIP NO PROFILE";; /* Tells to use Internal GateKeeper REGistration.../* ADD-GKREG:GWNO=811,GWATTR=EXTGW&HG3550V2&SIP,GWIPADDR=172.19.208.158, GWDIRNO=811,DIPLNUM=0,DPLN=0,LAUTH=1,INFO="XP_AUDIO_CONFERENCE_NO_SIP_PROFILE"; /* SIP Provider/Vendor/Device IP destination e.g.: 172.19.208.158 /*
You can use any routing type ( 4.1 and/or 4.2 and/or 4.3 ) adding on LDAT: GW1=... Example below using Route 4.2. Closed Number:
CHA-LDAT:LROUTE=83,LSVC=VCE,LRTEL=1,GW1=811-0; GW2 to 5 = alternate failover IP destination route (if GW1 broke, goes to GW2 ). +------------------------------------------------------------------------------+
The TDCSU: Protocol, Segment 2 (of 8) + LDAT GW/GKREG will send to the SIP GW a SETUP with Info Element: IE:
7
(56) : (1C)
QSIG facility 9F AA 06 80 01 00 82 01 00 8B 01 00 A1 2A 02 02 14 82 06 08 2B 0C 02 88 53 02 01 64 30 1A 30 0C Networking Extension Network Facility Extensions InvokeComp - ID: 1482 Component type: Global value Operation: Nq24 TargetGateways (100) targetList Gateway Addr: 172.19.208.158 Port: 5060
/* If need to delete the ldat lcr gkreg gw entry /*
DEL-LDAT:LRTEL,83,VCE,1,1;
For another Vendor IP Routing address using the same STMI, create another external GKREG (e.g. 812) and L CR.
9
4.4.2. SIP GW WBM change for STMI - Native SIP + GKREG: WBM => Explorers -> Voice Gateway -> SIP Trunk Profiles Parameter: Uncheck the field " Use Profiles for Trunks via Native SIP: " Apply => Save clicking on botton diskett icon => => Reboot clicking on circular arrow icon next to diskett
( see details onbotton on figure of item 5.1 )
For the next item 5, SIP with Profiles (no GKREG item 4.4) leave the field checked...
4.5. Miscelaneous AMO's system configuration for USA: CHANGE-CTIME:TYPESWU=CP2,GWNAVAIL=10;
/* GW Not AVAILable; 10 seconds configured, std =60 sec. /* blocktime if far ip is temporary out => F4066
F4066 M8 N0100 NO ACT BPA CP ADVISORY 14-04-16 15:58:40 ALARM CLASS:CENTRAL:023 CC:15362 EC: 2017 UA:B2C8:4E91 SP:B764:143C LD:01-11-001-000 DT:6C ST:6F SN: 0 CEVT:CD CSEV: F CST:15 Cha-ZAND:type=ALLDATA,CNTRYCD=K,SIUANN=D; /*=> for US slma, siu announcement CHANGE-ZAND:TYPE=ALLDATA,HOLDTN=RA, ANATESIG=RA; /* music on hold options CHANGE-ZAND:TYPE=ALLDATA,TRANSFER=EXTEND, EXCOCO=YES; /* transfer CO to CO calls CHANGE-ZAND:TYPE=ALLDATA2,USRINGTY="K"; CHANGE-ZAND:TYPE=ALLDATA2,LNR=YES,CAMPON=YES; CHANGE-ZAND:TYPE=CONFC,CODE=ULAW,CONFAT=3,STNAT=3; /*=> PCM24 uLaw. CHANGE-ZAND:TYPE=TN,SIUC="0K",DTR="0K",RDS="0K",CONFC="K"; /*=> US Tones CHANGE-ZAND:TYPE=TONES,CP=HOLDLINE,SIU="22"; /* internal Mozart music ACT-TREF:K; /* US Attenuation table COPY-PRODE:PVCD,16; /*=> US Trunk Protocols. CHANGE-ZAND:TYPE=OPTLOAD; /*=> Phone Protocols. CHANGE-ZAND:TYPE=ALLDATA2,TEXTSEL=AMERICAN&GERMAN&BRAZIL&SPANISH&ENGLISH; CHA-ZAND:TYPE=OPTISET,DISPLOGO=" UniFy-OS4k V7R1"; CHANGE-ZAND:TYPE=OPTISET,DISPTIME="12HOUR",ACOUSTID=USA,LANGID=ENGLISH&GERMAN&PORTUG&SPANISH; /* => for US digit.phones
10
4.6.WAYS TO
incoming CONVERT
DID NUMBERS:
W.1. DIDCR => TDCSU: TRTBL = didcr Example: External called 972.550-7150; Telco sends 4 digits 7150 to TDCSU.... Numbers to convert to Stations ( 87150 and 27010/37500) 7150 => 87150 7170 ==>27010 7180 ==>37500 DIDCR needs to receive 4 digits (7150) delete 1 (7) = 150 Take 15 and transforms to 8715 and use the last "0"; will cover stn 87150 to 87159... CHANGE-DIDCR:TYPE=SRT,SRTABLE=4,NETSVC=NONE,RULEIDX=6; /*tdcsu:DEDSVC=NONE, SRTIDX=4 DIS-DIDCR:SRT; PRI SERVICE RULE TABLE SRTABLE=4 NETWORK SERVICE RULEIDX ------------------------FX 0 INTLWATS 0 INWATS 0 LDS 0 NONE 6 PPC900 0 SDS 0 TTA 0 VPN 0 ADD-DIDCR:RULEIDX=6,TOTDIG=4,DELDIG=1; CHANGE-DIDCR:TYPE=DID,RULEIDX=6,TBLIDX=15,DIGITS=8715; CHANGE-DIDCR:TYPE=DID,RULEIDX=6,TBLIDX=17,DIGITS=2701; CHANGE-DIDCR:TYPE=DID,RULEIDX=6,TBLIDX=18,DIGITS=3750; DIS-DIDCR:RULETBL,6; -----------------------------------------------------DID DIGIT CONVERSION TABLE RULEIDX= 6 ------------------------------------------------------TOTAL NUMBER OF EXPECTED INCOMING DIGITS: 4 NUMBER OF DIGITS TO BE DELETED: 1 TBL_ TBL_ TBL_ TBL_ TBL_ IDX DIGITS IDX DIGITS IDX DIGITS IDX DIGITS IDX DIGITS ------------------------------------------------------------------------------00 20 40 60 80 01 21 41 61 81 02 22 42 62 82 11 31 51 71 871 91 12 32 52 72 92 13 33 53 73 93 14 34 54 74 94 15 8715 35 55 75 95 16 36 56 76 96 17 2701 37 57 77 97 18 3750 38 58 78 98 19 39 59 79 99 ------------------------------------------------------------------------------/* If the convertion is only for 71xx to 871xx then: /* DIDCR receives 4 digits (71xx; Number of Digts to be Deleted: 0 (zero) /* Take TBL_IDX 71 and transforms to DIGITS 871; will cover stns. 87100 to 87199... ========================================================================================== W.2.TDCSU: TRTBL = GDTR What Telco sends will be accept. Can be manipulated further with TDCSU: SUPPRESS and/or DGTPR or RUFUM
W.3. RUFUM (SYSTEM Wide for TRTBL=GDTR to all DID trunk types)..
+-----------+-----------------------------+--------------+ | TYPE | DIALLED CODE | CONV. CODE | +-----------+-----------------------------+--------------+ | DID | 7150 | 87150 |
W.4. Insert # on TDCSU: DGTPR=8 SUPRESS=0 CHA-TDCSU:PEN=1-81-2-0,DGTPR=8; External caller dialed 972.550-7157 Telco PRI T1 NI2 will sent to TDCSU 7157 TDCSU will add 8 in front of 7157 to call the physical station 87157. TO DELETE THE TDCSU:DGTPR;
USE
CHA-TDCSU: PEN=X-Y-Z-W, DGTPR= * ;
/* ONE STAR "*" to the hell
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5. VoIP = SIP GW. Web Browser Management (WBM) Settings ; inside STMI card. 5.1. Activation of SIP Trunk Profile GW: Open OS4k - Assistant -> Expert Mode -> Gateway Dashboard -> HG35xx Web Base Management (WBM) on WBM, update board list and connect to the STMI card (example: PEN = 1-1-11) WBM => Explorers -> Voice Gateway -> SIP Trunk Profiles (Choose one that is appropriate for the Sip Vendor/Provider connection ..) => right mouse click on e.g. NativeTrunkWithoutRegistration -> Enter the Native SIP IP address of the destination provider/vendor/device on Proxy (e.g.: 172.19.208.158); then scroll down on the page and click the Apply button. Then again; right mouse click on NatTrkWithoutRegistration -> Activate
Folder; turns green =>
Ask SIP Provider/Vendor which transport protocol is using (TCP or UPD)... Check below fields always, after card replaced !!!!
Unlocked for write access !
Finally Click on Diskette icon to save it....
|| running lw pzksti40.A1.....
Continuation of SIP Trunk Profiles List on V7R0 lw pzksti40.A1.002-008 at STMI card:
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For virtual STMI (vSTMI ) running on SoftGate (SG lw pzksgw50.A1.007-007) there are 3 more profiles choices: + Microsoft - Lync + Microsoft - OCS + XPressions => (native Sip, but this connection lose the display messages from XP) - The list can be different with less or more providers in old or future lw version; do always consult / display the running STMI with WBM => Explorers -> Voice Gateway -> SIP Trunk Profiles ...
From GVS Hot Issue Notif. #31 June-2015: rd Item 9. Correct HG3500 SIP Trunk Profile to be used for 3 Party Applications
There is a common misunderstanding with the usage of SIP Trunk Profiles used on the HG3500 and vHG3500 boards. When no specific named rd profile exists for 3 Party Applications (not to be confused with SIP Service Provider), meaning the application is not explicitly certified, then the best profile to use is NatTrkEnterprise. NatTrkEnterprise as per the WBM Profiles Details is described as “Recommended Profile for 3rd Party SIP Applications using REFER”. This profile is often confused with either NatTrkWithRegistration or NatTrkWithoutRegistration which were intended for SIP Service Providers. Certification for SIP service providers is still a mandatory requirement as per official Sales Information literature . Moreover please be aware the current released SIP Service Provider list is linked from Release Notes and can be found under INF-13-000534 .
Ping the far end:
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5.2. Other possible SIP GW settings :
Figure 5.2.1
Figure 5.2.2
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WBM=> Explorers -> Payload => right mouse click on HW Modules
Figure 5.2.3
DTMF Outband Sign.= only for H323 connection (e.g. XPressions); not for SIP (RFC2833 or inband)
Figure 5.2.4
Figure 5.2.5
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Figure 5.2.6
Figure 5.2.7
Figure 5.2.8
Figure 5.2.9
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6.
T R O U B L E S H O T I N G :
Stn. <===> OS4k /HP4k -- rmx/tdm -- STMI/SIP-GW <=== VoIP Lan Network ===> SIP Provider/Vendor 172.19.208.184 172.19.208.158
6.1. Wireshark ( Network Sniffer Trace) on VoIP Lan Network; first most efficient break point. Mirror the STMI Lan Network port, on L2 Switch and use Wireshark to Capture -> Interfaces -> select the PC L an interface port -=> Start do ping from STMI to see if you're capturing frames; if so then make test call from a Stn. (e.g. stn.7150 dialing 811, routing 4.3. used ) towards SIP Provider and stop capture (red square) - Choose Filter " ip.addr==172.19.208.184 " --> Apply Another good Filter " SIP " --> Apply - If Invite goes out, OS4k / STMI looks ok. - If you don't see the answer from Destination IP.158, contact the SIP Vendor/Provider IP. - If you don't see the Invite go ing out; then it's OS4k internal problem on rmx or stmi gw...
For details, right click on - Session Initiation Protocol (INVITE), on above midle window => select Expand Subtrees
If you select on top main menu: Telephony --> VoIP Calls => display all calls on a capture file If want to hear G711 conversation, click Player ( listen discretion advised) Highlight one or all call and click Flow
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Wireshark, Native Sip with TCP as signaling transport protocol (instead upd).
Example below shows a test call from Stn. 7152 called, talked (RTP G711) to 811 Native Sip connection. Note that the SIP Provider does the signalling on IP 172.19.208.158 and the Media Server (audio) is going on IP .174
- Wireshark SW can be download at http://www.wireshark.org/download.html
it's free :-)
- Incoming sip call, all channels b usy = sip message 503 Service Not Available. (see item 7. Message conversion) - Outgoing call, all channels busy = station get "Please Try Later".
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6.2. RTDS, internal TDCSU ISDN trace; Second Break Point. Assistant -> Diagnostics -> Switch Diagnosis System -> Real Time Diagnosis System => Click on Monitor Trunk / ISDN Trace Choose the TDCSU pen, Type=ISDN Tracer and Start
Login with any charachter...
Click on ISDN Trace Display and make a test call (e.g.: stn.7166 dialing 811) If setup is going out, then rmx/amos are ok.... If not check stn. trk config. and do a rmx-trace, item 6.4. If setup is going out, but disconnect comes right after, seems a sip gw problem; do a stmi-gw trace i tem 6.5.
Click on Message Type, e.g. SETUP:
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6.3. Maintenance AMO commands: /* Status Display in Switching Unit /* PEN LTG=1- LTU=1- SLOT=11 NO PKI CERTIFICATE AVAILABLE LINK SIGNAL ETHERNET . . . . PRESENT LAN SPEED . . . . . . . . . 100 MBIT/S LAN INTERFACE . . . . . . . . FDX (FULL DUPLEX) => Lan connection is running 100 MBFD... CCT FUNCTION BLOCK 0 - 1 HG3550_2 2 - 52 HG3530_2 53 - 60 HG3540_2 CCT LINE STNO SI BUS TYPE 000 2160 PP NW READY => Circuit 0 of TDCSU pen 1-1-11-0 MULTLINE 30 . . . . . . . . . . . . . .READY 000 NO CONN 001 NETWORK SUBUNIT . TMD CONN ISDN READY (ALT_ROUT: N) (HG3550IP) LINE: 2160 STNO: SI: 001 . . . . . . . . TMD CONN ISDN READY 002 NETWORK SUBUNIT . TMD CONN ISDN
=> Channel 1 = OK
READY/BUSY/CP CPH
=> Channel 2 in talk state
(ALT_ROUT: N) (HG3550IP) LINE: 2160 STNO: SI: 001 . . . . . . . . TMD CONN ISDN READY/BUSY ------------------------------------------------------------------------
/* Realtime Trunk Group Occupation /*
DIS-BUEND:11,B; NO. 11
MAX 10
ACT 10
BUSY FREE
3
7
OTHER 0
/* Reset Device Service BChannel /*
RES-DSSU:BCHAN,1-1-11-0,1&&30; /* Reset Device Trk (tdcsu pen) /*
RES-DSSU:PEN,1-1-11-0,;
/* Reset the Board (card) Service/*
RES-BSSU:PEN,1,1,11; /* RMX Switching Unit Reset /*
soft, hard or reload...
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Realtime GW ports occupation (e.g. 3 calls):
6.4. RMX - Traces: - Do it if you don't see the Setup going out or the RTDS is temporary not working. This covers stn. to trk messages (rmx & isdn) until the SIP GateWay. cha-diags:comp=cp,s14=on; ex-tracs:bp; res,all; flagtr,off; selmsg,pp,g1,cd1,ev,ba&97&74; /* Standard PP networking selmsg,pp,g1,cd2,byte,15,08; msglen,pp,g1,280; selmsg,pp,g2,all; /* All other PP messages msglen,pp,g2,32; selmsg,cp,g2,all; /* All other CP messages msglen,cp,g2,48; selmsg,rcv,g1,cd1,dest,40; /* FA TASK selmsg,rcv,g1,cd2,src,40,ne; on,hd,:diag:,30,y,y; /* = e.g. nsip1 /* do the test call off; Utilize ComWin, File Transfer and download the :diag: To read it need the Tracex Tool powered by Mike BJ or the MessageDoctor SW by GVS.
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6.5. STMI - GW Traces: - Do it if ISDN message (e.g. Setup) is going towards STMI, but you don't see an Invite on Wireshark going out . XTracer SW Instructions: x1. To get the XTRACER SW go on Unify Intranet: http://prodsu.mch4.global-intra.net/ => Tools/Links or http://cur1303x.global-ad.net/foswiki/pub/HUSIM/XTracerDownload/setup.exe
x2.Access Web Based Management and connect to the Gateway. x3. Connect to the STMI/HG35xx Card, then on Maintenance -> Traces -> --> Trace Profiles => right mouse click on 1.1.2 SIP trk. General probl. and Activate Profile
x4. --> Scroll all the way to the top and right click on Trace Output Interfaces and click on Edit Trace Output Interfaces
--> Uncheck everything and check Switch Trace via LAN On, click on Apply button --> Verify on Trace Format Configuration if all Header Output choices are set to Yes, and the Trace Data is Full formatting with Parameter Expansion. ==> Click on the diskette icon to Save. - If you're not able to get XTracer (Dev.request), choose Switch File Trace On and collect together o n item x8. x5 .Launch XTracer -> click on the lightning bolt, On the Start Tab select Trace Type of “HiPath2000 LDH (FP inside)…” and add you IP address in the IP address field. Further windows asking Decoders, select NONE. At the bottom, add where you want your file to be saved. Click Start and do the test or wait until fail occurs.... x6.Can be saved in multiple files cli ck on the Ring Buffer Tab (on Start Trace Tabs) Next File Every (size) ****** The maximum file size of one t race is 1.99Gb. Can also start another trace/card in parallel (change the Tracing File destination name) x7. When finished stop the trace as this overload the STMI processor. Maintenance->Traces->Trace Components => right mouse click -> Stop All Trace Components and click on the diskette icon to Save. Uncheck Switch Trace via LAN on Trace Output Interfaces. x8. Collect all STMI card logs with: Maintenance->Actions->Manual Actions->All logs=> right click -> Load via HTP -> Load Save on your PC.
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6.6. STMI - Loadware ( LW ). - LW is the Operational System SW ( including the SIP stack ) that runs inside the card on STMI/NCUI Flash Memory (FM). - STMI LW is stored on the rmx HD area :PDS:APSP/LTG/LG98/PZKSTI40 - New LW can be consulted on SWS ( Unify - SoftWare Supply Server ), LW Hot Fix. - How to replace the LW; see the doc. SWS and Hotfix G uide, file HP4k_SWS_HF_Guide.pdf - To load a new STMI LW, takes about 13 minutes and card will be READY.
- When STMI/NCUI card resets/reboot/power off-on/pull-push, there is a card's flash memory (FM) check against to the lw stored on HD area :PDS:...., if the lw is the same (FM = HD), card boots in about 90 seconds... If not STMI takes 13 minutes to load new LW from HD .
Condition for that check: ----------------------------------------------------------------------------------------------------------------Card Loadware load behaviour on V6R1 down versions: /* return the funsu function to standard value; NO forced lw load always.... ------------------------------------------------------------------------------------------------------------------------------------------------Card Loadware load behaviour on V6R2 up: /* on Hard Restart, no forced lw load from hd. LWSRC: => /* LW source for Card Flash Memory (FM) load after card reset ========================== PARTNO FCTID LOADDEV ------ ----- ----Q2324-X500 1 HD => /* Force load lw from HD to the card (always 13 min.to STMI=ready; no FM=HD check). Q2324-X510 1 FLASH => /* check first if flash memory lw = HD lw; if not load from HD. AFTER HARDRESTART / RELOAD, VALUES FOR HRBHV & LWSRC WILL BE SET BACK TO DEFAULT NOLOAD / FLASH This amo zande, lw behaviour does NOT apply for NCUI or SoftGate...
----------------------------------------------------------------------------------------------------------------------------------------------------------!! V7 Exception on the condition above; no LW FM=HD check at all with ZANDE: NOIPBLWL= YES. Quick, fast IN Service (READY) !! New on V7 AMO- ZANDE: ALLDATA2, NOIPBLWL = yes ; /* If yes means NO IP boards lw load from hd. You get quicker reload as stmi/ncui loads from flash (STMI=90 sec. no hd check); but be aware that you should later change noipblwl=NO and reboot the stmi/ncui cards manually to get newest lw from the new replaced/upgraded HD... Summary: NOIPBLWL = yes; => STMI/NCUI will NEVER load LW from HD. STMI ready in 90 sec. with old/exist LW from card Flash Memory . NOIPBLWL = no; => behaviour is equal as on V6, V5. described as above on Card LW load topic.. ----------------------------------------------------------------------------------------------------------------------------------------------- --------------------------------
6.6.1. Ways to check running LW: - See on WBM, at the botton of figure from the item 5.1
or more info is on same WBM-. Explorers -> Basic Set. -> SW Build
or also go to Assistant -> Expert Mode -> Gateway Manager ... => "Update Board List" - ComWin rmx-amo I386 pzksti40.A2.001-005.00ÙËpzksti40.. 0040 :............h.......X}‡.................ðÿ...@..ªUªU........ ..ôÿ
=> /* compiled on April/09/2014 @ 14:19:21 /* lw version PZKSTI40.A2.001-005
/* card SLMAE current running lw pzesla40 01|002|1|:PDS:APSP/LTG/LG83/PZDSMO10 |--|12/19/12 17:10:47| => /* SLMO24 on pen 1-2-1 pzdsmo10 lw, Dec.19, 2012 01|011|1|:PDS:APSP/LTG/LG98/PZKSTI40 |00|04/09/14 14:19:21| => /* ltu=1 slot=11 STMI lw from April/09/2014
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6.7. STMI - CLI = Command Line Interface. 6.7.1. OS4k - Assistant -> Expert Mode -> Gateway Dashboard => SSH = CLI
6.7.2. Can get CLI also from your PC with serial null cable DB9 to the STMI - Service Port: 38400 N-8-1 On this serial connection, you can follow the booting messages when card is loading.... 6.7.3. Login with username: TRM password: HICOM If this not work do < CHANGE-CGWB:MTYPE=CGW,LTU=1,SLOT=11,TYPE=SERVIF,LOGINTRM="TRM",PASSW=HICOM; 6.7.4. Some intersting commands..
Welcome to the HG 3500 V7 pzksti40.A2.001-005 Command Line Interpreter. vxTarget> help
=> shows all possible commands
get write access get id eth_link_mode => should be 100M Full Duplex ping reset reset factory save configuration switch console trace on show active traces switch console trace off reset active traces show show show show show show show show show show show show show show show
=> shows the trace in real time; only on serial cable DB9 connection
arp cache boot default gateway flash => directories/files if counters => dropped frames if states => if = interfaces ip address memory mscindexes => number of call connections routes time uptime versions => running lw version cpu usage file_ascii
Ctrl W logout
=> goes to debug/development mode
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7. Default SIP -to- SS7 ISUP ( ISDN ) Cause Codes : ISUP (TDCSU) Cause Value - SIP (STMI GW) Response (400 ~ 699 )
1 – unallocated number - 404 Not Found 2 – no route to network - 404 Not Found 3 – no route to destination - 404 Not Found 16 – normal call clearing - --- (*) 17 – user busy - 486 Busy here 18 – no user responding - 408 Request Timeout 19 – no answer from the user - 480 Temporarily unavailable 20 – subscriber absent - 480 Temporarily unavailable 21 – call rejected - 403 Forbidden (+) 22 – number changed (s/ o diagnostic) - 410 Gone 22 – number changed ( w/ diagnostic) - 301 Moved permanently 23 – redirection to new destination - 410 Gone 26 – non-selected user clearing - 404 Not Found (=) 27 – destination out of order - 502 Bad Gateway 28 – address incomplete - 484 Address incomplete 29 – facility rejected - 510 Not implemented 31 – normal unspecified - 480 Temporarily unavailable Resource unavailable 34 – no circuit available - 503 Service unavailable 38 – network out of order - 503 Service unavailable 41 – temporary failure - 503 Service unavailable 42 – switching equipment congestion - 503 S ervice unavailable 47 – resource unavailable - 503 Service unavailable Service or option not available 55 – incoming calls barred within CUG - 403 Forbidden 57 – bearer capability not authorized - 403 Forbidden 58 – bearer capability not presently available - 503 Service unavailable 65 – bearer capability not implemented - 488 Not Acceptable here 70 – Only restricted digital information bearer capability is available (National use) - 488 Not Acceptable here 79 – service or option not implemented - 501 Not implemented 87 – user not member of CUG - 403 Forbidden 88 – incompatible destination - 503 Service unavailable 95 - Invalid message - 400 Bad request 102 – Call Setup Time-out Failure - 504 Gateway timeout 111 – Protocol Error Unspecified - 500 Server internal error Interworking 127 – Internal Error - interworking unspecified - 500 Server internal error - - - 500 – Server internal error (default)
SIP Status Code to ISDN Cause Code Mapping
400 – Bad Request 41 – Temporary failure 401 – Unauthorized 21 – Call rejected (*) 402 – Payment required 21 – Call rejected 403 – Forbidden 21 – Call rejected 404 – Not Found 1 – Unallocated number 405 – Method not allowed 63 – Service or option unavailable 406 – Not acceptable 79 – Service/option not implemented (+) 407 – Proxy authentication required 21 – Call rejected (*) 408 – Request timeout 102 – Recovery on timer expiry 409 - Conflict 41 - Temporary failure 410 – Gone 22 – Number changed (w/o diagnostic) 411 - Length required 127 - Interworking, unspecified
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413 – Request Entity too long 127 – Interworking (+) 414 – Request –URI too long 127 – Interworking (+) 415 – Unsupported media type 79 – Service/option not implemented (+) 416 – Unsupported URI Scheme 127 – Interworking (+) 402 – Bad extension 127 – Interworking (+) 420 - Bad extension 127 - Interworking, unspecified 421 – Extension Required 127 – Interworking (+) 423 – Interval Too Brief 127 – Interworking (+) 480 – Temporarily unavailable 18 – No user responding 481 – Call/Transaction Does not Exist 41 – Temporary Failure 482 – Loop Detected 25 – Exchange – routing error 483 – Too many hops 25 – Exchange – routing error 484 – Address incomplete 28 – Invalid Number Format (+) 485 – Ambiguous 1 – Unallocated number 486 – Busy here 17 – User Busy 487 – Request Terminated --- (no mapping) 488 – Not Acceptable here --- by warning header 500 – Server internal error 41 – Temporary Failure 501 – Not implemented 79 – Not implemented, unspecified 502 – Bad gateway 38 – Network out of order 503 – Service unavailable 41 – Temporary Failure 504 – Service time-out 102 – Recovery on timer expiry 505 – Version Not supported 127 – Interworking (+) 513 – Message Too Large 127 – Interworking (+) 580 - Precondition Failed 47 - Resource unavailable, unspecified 600 – Busy everywhere 17 – User busy 603 – Decline 21 – Call rejected 604 – Does not exist anywhere 1 – Unallocated number 606 – Not acceptable --- by warning header (*) ISDN Cause 16 will usually result in a BYE or CANCEL (+) If the cause location is ‘user’ than the 6xx code could be given rather than the 4xx code. the cause value received in the H.225.0 message is unknown in ISUP, the unspecified cause value of the class is sent. (=) ANSI procedure
The information provided in this document contains descriptions or characteristi cs of performance which in case of actual use do not always apply as described or which may change as a result of further development of the products. Availability and technical specifications are subject to change without notice. An obligation to provide the respective charact eristics or support and warranty shall only exist if expressly agreed in the terms of contract. Neither the author nor Unify shall be liable for errors, omissions or damages resulting from the use of the informations contained herein.. The comments expressed belong to the author and are not necessarily those of Unify Inc. OpenScape, OpenStage and HiPath are registered trademarks of Unify. All other company, brand, product and service names are trademarks or registered trademarks of their respective holders.
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