Call Transfer issue, No audio and no voice

[rpvish2004]

Hi

we have Nortel CS1000E 7.6 system which is integrated with CISCO CUCM via Session manager SIP Trunk.
I have created DSC for CISCO Extn And routing has been done in Session manager.
The integration is successful Nortel Phone is able to call Cisco and Vice versa happening.
But is issue is with call transfer
eg Nortel IP Phone Extn 50133 called to Cisco Extn 50003 Voice is OK
Cisco Extn 50003 Transfer The same call to Another Nortel IP phone Extn 50572
After the call getting transferred there is no Audio/Blank between both Nortel IP Extn 50133 and 50572.

Here is the DCH Configure, please suggest what need to be done at Avaya end or CISCO end.

ADAN DCH 1
CTYP DCIP
DES VDCH
USR ISLD
ISLM 4000
SSRC 1800
OTBF 32
NASA NO
IFC SL1
CNEG 1
RLS ID 5
RCAP ND2 MWI
MBGA NO
H323
OVLR NO
OVLS NO

 

[Firebird Scrambler]

I'm not sure if you need to have "TRMB" set to "YES" against the routes RDB in LD 16.?

Firebird Scrambler

Nortel & Avaya Meridian 1 / Succession & BCM / Norstar Programmer

Website =

http://somertel.net

linkedin

 

[rpvish2004]

Thanks Sir for your revert.

I checked the SIP rout 'TRMB' is already set to YES.

here is the print of Route

TYPE RDB
CUST 00
ROUT 103
DES SIP_TRUNK
TKTP TIE
NPID_TBL_NUM 0
ESN NO
RPA NO
CNVT NO
SAT NO
IDEF NET
RCLS EXT
VTRK YES
ZONE 00003
PCID SIP
CRID YES
SBWM NO
NODE 1000
DTRK NO
ISDN YES
MODE ISLD
DCH 1
IFC SL1
PNI 00001
NCNA YES
NCRD YES
TRO NO
FALT NO
CTYP UKWN
INAC NO
ISAR NO
DAPC NO
MBXR NO
MBXOT NPA
MBXT 0
PTYP ATT
CNDP UKWN
AUTO NO
DNIS NO
DCDR NO
ICOG IAO
SRCH LIN
TRMB YES
STEP
ACOD 7103
TCPP NO
PII NO
AUXP NO
TARG 01
CLEN 1
BILN NO
OABS
INST
IDC NO
DCNO 0 *
NDNO 0
DEXT NO
ANTK
SIGO STD
STYP SDAT
MFC NO
ICIS YES
OGIS YES


PAGE 002

PTUT 0
TIMR ICF 512
OGF 512
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 ETH
FEDC ETH
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 1
FRL 2 2
FRL 3 3
FRL 4 4
FRL 5 5
FRL 6 6
FRL 7 7
AUTH NO
TTBL 1
ATAN NO
OHTD NO
PLEV 2
OPR NO
ALRM NO
ART 0
PECL NO
DCTI 0
TIDY 7103 103
ATRR NO
TRRL NO
SGRP 0
ARDN NO
CTBL 0
AACR NO

 

[Firebird Scrambler]

The only other thing I can think of is if there is another setting in "RCAP" that needs to be looked at.

Firebird Scrambler

Nortel & Avaya Meridian 1 / Succession & BCM / Norstar Programmer

Website =

http://somertel.net

linkedin

 

[rpvish2004]

Please suggest what option I could enable for this issue as currently SIP DCH has RCAP ND2 MWI.

And all external calls are going through with this DCH, we have to take downtime for any changes in SIP DCH.

As we are facing issue only while call getting transferred from CISCO Extn.

Nortel calls---to CISCO-----Transferred call-----to Nortel ===> No audio/Blank

Nortel to Nortel and CISCO to CISCO the issue is not happening.

Also please suggest how we can disable/enable the SIP DCH for making any changes.

 

[AvayaHE]

We have the same configuration active.
In contrast to you, we do not use standard in the routing script ( SIGO = STD) and in the D channel RCAP = TAT ( anti-trombon )
We operate the trunks with wnk/wnk instead of IMM/IMM .
most often such problems occur in connection with several features.
A standard SIP trunk is installed in the Cisco system.
I have attached a few more screenshuts .
Good luck

ADAN DCH 15
CTYP DCIP
DES VDCH
USR ISLD
ISLM 4000
SSRC 1800
OTBF 32
NASA YES
IFC SL1
CNEG 1
RLS ID 4
RCAP MWI TAT ND3
MBGA NO
H323
OVLR NO
OVLS NO

RDB
TKTP TIE
NPID_TBL_NUM 0
ESN NO
RPA NO
CNVT NO
SAT NO
IDEF NET
RCLS INT
VTRK YES
ZONE 00017
PCID SIP
CRID NO
SBWM NO
NODE 1010
DTRK NO
ISDN YES
MODE ISLD
DCH 15
IFC SL1
PNI 00001
NCNA YES
NCRD NO
FALT NO
CTYP UKWN
INAC NO
ISAR NO
DAPC NO
MBXR NO
MBXOT NPA
MBXT 0
PTYP ATT
CNDP UKWN
AUTO NO
DNIS NO
DCDR NO
ICOG IAO
SRCH LIN
TRMB YES
STEP
ACOD 110517
TCPP NO
PII NO
AUXP NO
TARG
CLEN 1
BILN NO
OABS
INST
IDC NO
DCNO 0 *
NDNO 0
DEXT NO
SIGO ESN5
MFC NO
ICIS YES
OGIS YES
TIMR ICF 512
OGF 512
EOD 13824


PAGE 002

DSI 34944
NRD 10112
DDL 70
ODT 4096
RGV 640
GTO 896
GTI 896
SFB 3
NBS 2048
NBL 4096

IENB 5
TFD 0
VSS 0
VGD 6
EESD 1024
SST 5 0
DTD NO
SCDT NO
2 DT NO
NEDC ETH
FEDC ETH
CPDC NO
DLTN NO
HOLD 00 00 0
SEIZ 00 00
SVFL 00 00
DRNG NO
CDR NO
NATL YES
SSL
CFWR NO
IDOP NO
VRAT NO
MUS NO
PANS YES
MANO NO
FRL 0 11
FRL 1 11
FRL 2 11
FRL 3 11
FRL 4 11
FRL 5 11
FRL 6 11
FRL 7 11
OHQ NO
OHQT 00
CBQ NO
AUTH NO
TTBL 0
ATAN NO
OHTD NO
PLEV 2
OPR NO
ALRM NO
ART 0
PECL N

TIE

DES SIP
TN 112 1 09 29 VIRTUAL
TYPE IPTI
CDEN 8D
CUST 0
XTRK VTRK
ZONE 00017
LDOP BOP
TIMP 600
BIMP 600
AUTO_BIMP NO
NMUS NO
TRK ANLG
NCOS 7
RTMB 17 1340
CHID 1800
TGAR 8
STRI/STRO WNK WNK
SUPN YES
AST NO
IAPG 0
CLS UNR DTN CND ECD WTA LPR APN THFD XREP SPCD MSNV
P10 NTC
TKID
AACR NO

 

[rpvish2004]

Thanks for your revert.

I will try to add RCAP to TAT.

But for that we need to disable SIP DCH,

Please suggest me how to disable the SIP DCH, will take the downtime accordingly.

 

[rpvish2004]

Hi Team,

Please provide help to resolve the issue.

Can Changing RCAP to TAT will resolve the issue ? as we will not get the downtime again.

Any other setting need to changed. As this issue occurs when the call getting transferred from CISCO Extn.
Is there any setting needs to checked at CISCO end.

Just to update you all,
1) Nortel Cs1000e system and CISCO System is connected Via Session Manager.
2) CISCO Extn is created as DSC in CS1000e.

In the information is enough for resolution, Kindly revert.

 

[AvayaHE]

Sorry ,was out of the office
Disable D-Chanel has be done overlay 96
Ld 96 Enter
dis dch xx Enter
Then open overlay 17 to made your change
Return
Overlay 96
enl dch xx Enter

 

[rpvish2004]

Thanks For your revert.

Please let me is there any more setting need to be checked.

As I have enabled TAT in DCH but the issue is not resolved.

Anyone one please help to resolve this issue.

 

[AvayaHE]

I am Back
Not so easy to say what else
Please check configuration of CISCo System

Region Configuration

Check Audio Bit Rate
Must be same Audio Codec as Avaya Sessionmanager and Signalinhserver
G711 G729 or what ever

SIP Trunk Security Profile

Incoming Transport Type Use UDP and TCP ( non Secure )
Outgoing TCP ( or what you set Avaya Session Manager )
Port 5060

Activate
Accept out-of-dialog refer**
Accept unsolicidet notification
Accept repleace headwer


In SIP Profile ( leve ost default )

Default MTP Telefonie Event Payload Type = 101
SDP Session-level Bandwith Modifier for Early Offer and Re-Invite = TIAS and AS

Activate
Disable Early Media on 180


Scroll to SIP Information section, enter the following values and use defaults for remaining
fields:
Destination Address Enter IP address of SIP signaling interface for Session Manager.
Destination Port Defaults to “5060”.
MTP Preferred Originating Codec Select “711ulaw or your local Codec in use ”.
SIP Trunk Security Profile Select SIP Trunk Security Profile .
SIP Profile Select SIP Profile
DTMF Signaling Method Select “RFC 2833”.