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Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

(OP)
Hi all,

I have configured Polycom Trio SIP Phone as SIP 3party on the CS1K system and it is registered correclty:

Incoming calls (From CS1K Unistim phone to Polycom SIP Phone) are working fine
When we call from SIP Phone to Unistim phone, I got busy tone, and on the traces I can see 407 authentification required returned by CS1K followed by 403 Forbidden

When we redial from missed call of the SIP Phone (Polycom) to call Unistim phone, it works fine !!

In my investigation, when I compare bad and good calls, I can see that the good calls (redial from missed calls list) sent the INVITE as follow:

"sip:1000dialsrc=calllistBphone-contextdialsrc=calllistDcdp.udp@mydomain"

Bad call (normal outbound call) with INVITE like: "sip:1000@mydomain;user=phone"

I guess that the CS1K don't accept INVITE format without "phone-context=cdp.udp"

Please any idea of how to put adaptation or something on the CS1K side ?

Thank you.

RE: Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

You would have to use NRS or Session Manager which ever you have to change the call to the correct format. For instance I have a Session manager and I connect calls between my CS1K and my Avaya CM. I have an adaption in session manager that changes the outgoing source id so they play nice together.

RE: Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

(OP)
Hi tbonz25,

Thanks for your reply, in my case, I don't have Session Manager, but I have NRS

I don't believe will need NRS for SIP Line phones

If so, how we can configure SIP adaptation on that NRS ?

Thank you.

RE: Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

NRS is not need for SIPL Application
SIPL is a NODE communication Solution
Must be someting wrong on your Routing ,Trunk or DCH
Can you present those script ?

RE: Add adaptation to CS1000 from sip:1xxx@mydomain;user=phone to "phone-context=cdp.udp"

(OP)
Hi AvayaHE,

Here is the configuration of DCH:

TYPE adan dch 100

ADAN DCH 100
CTYP DCIP
DES SIPL
USR ISLD
ISLM 4000
SSRC 3700
OTBF 32
NASA NO
IFC SL1
CNEG 1
RLS ID 25
RCAP ND2
MBGA NO
H323
OVLR NO
OVLS NO

============
RDB configuration:

ROUT 200

TYPE RDB
CUST 00
ROUT 200
DES SIPL
TKTP TIE
NPID_TBL_NUM 0
ESN NO
RPA NO
CNVT NO
SAT NO
RCLS EXT
VTRK YES
ZONE 00001
PCID SIPL
CRID NO
NODE 100
DTRK NO
ISDN YES
MODE ISLD
DCH 100
IFC SL1
PNI 00001
NCNA YES
NCRD YES
TRO NO
FALT NO
CTYP CDP
INAC NO
ISAR NO
DAPC NO
MBXR NO
MBXOT NPA
MBXT 0
PTYP ATT
CNDP CDP
AUTO NO
DNIS NO
DCDR NO
ICOG IAO
SRCH LIN
TRMB YES
STEP
ACOD 68010
TCPP NO
PII NO
AUXP NO
TARG
CLEN 1
BILN NO
OABS
INST
IDC YES
DCNO 3
NDNO 3 *
DEXT NO
DNAM NO
SIGO STD
STYP SDAT
MFC NO
ICIS YES
OGIS YES
TIMR ICF 512
OGF 512


PAGE 002

EOD 13952
DSI 34944
NRD 10112
DDL 70
ODT 4096
RGV 640
GTO 896
GTI 896
SFB 3
NBS 2048
NBL 4096
TFD 0
EESD 1024
SST 5 0
DTD NO
SCDT NO
2 DT NO
NEDC ORG
FEDC ORG
CPDC NO
DLTN NO
HOLD 02 02 40
SEIZ 02 02
SVFL 02 02
DRNG NO
CDR NO
NATL YES
SSL
CFWR NO
IDOP NO
MUS NO
PANS YES
RACD NO
MANO NO
FRL 0 0
FRL 1 0
FRL 2 0
FRL 3 0
FRL 4 0
FRL 5 0
FRL 6 0
FRL 7 0
AUTH NO
TTBL 0
ATAN NO
OHTD NO
OPR NO
ALRM NO
ART 0
PECL NO
DCTI 0
TIDY 68010 200
ATRR NO
TRRL NO
SGRP 0
ARDN NO
CTBL 0
AACR NO

==================================

Trunk members:


DES SIPL
TN 100 0 00 00 VIRTUAL
TYPE IPTI
CDEN 8D
CUST 0
XTRK VTRK
ZONE 00001
LDOP BOP
TIMP 600
BIMP 600
AUTO_BIMP NO
NMUS NO
TRK ANLG
NCOS 7
RTMB 200 1
CHID 1
TGAR 1
STRI/STRO IMM IMM
SUPN YES
AST NO
IAPG 0
CLS UNR DIP CND ECD WTA LPR APN THFD XREP SPCD MSBT
P10 NTC MID
TKID
AACR NO
DATE 26 DEC 2018


Thanks for your time

Regards.


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