Third party SIP Phone unable to register on the SLG of CS1000

[nortavaya]

Hi all,

We try to register a third party SIP Phone to the CS1K (through SLG)but with no succes, we did all required configuration on the CS1K side and SIP Phone side as follow:

1. Enable a SIP Third Party Licence on the CS1K System
2. Enable SLG on the Node IP
3. Configure SLG settings on the NodeIP (SIP Domain, SLG port, MO IP address, TCP...etc)
4. Create DCH,Zone, RDB and members
5. Use the existing AML 16 and VAS 16 (they are already used for AACC and enabled, can we use the same, or I should create new VAS/AML ?)
6. Create SIPL phone as UEXT type

Then, when I try to register the SIP Phone, I got error on the phone saying failed to register

How I can check if my SLG is up and running ? note that we have Signaling Server Linux based

Also, how I can enable realtime SIP Traces if SLG in order to see what happens and SIP messages exchanges ?


Thank you in advance

 

[tbonz25]

You have to create a new AML and VAS link. This has to be a number 32 or higher but I just use 32 for the AML and VAS. Also you must make sure that the AML link comes up as that is how the SIP line talks to any SIP phone third party or not.

 

[nortavaya]

Hi tbonz25,

Thanks for the reply, yes it working fine now,the SIP 3Party phone is register, however I have another issue when the sip phone try to dial out (either call internal phones, or external numbers), we got busy tone, but the incoming calls working fine
When I check the logs on the SIP Phone, we see that there is error "SIP/2.0 403 Forbidden" returned from SS and ''SIP/2.0 407 Proxy Authentication Required''

I alreay set CLS to UNR for UEXT phone and IPTI trunk, I checked also for the TGAR, everything seems correct

Please any idea what can be the issue ?

Thank you.

 

[tbonz25]

Can you turn on the dchannel messaging for the call on the virtual dchannel. Also send your programming of the UEXT and we can take a look.

 

[nortavaya]

>ld 20

PT0000
REQ: prt
TYPE: tn
TYPE TNB
TN 252 0 2 0
DATE
PAGE
DES

DES SIPMB5
TN 252 0 02 00 VIRTUAL
TYPE UEXT
CDEN 8D
CTYP XDLC
CUST 0
UXTY SIPL
MCCL YES
SIPN 0
SIP3 1
FMCL 0
TLSV 0
SIPU 58117
NDID 100
SUPR NO
UXID
NUID
NHTN
CFG_ZONE 00002
CUR_ZONE 00002
MRT
ERL 0
ECL 0
FDN
TGAR 1
LDN NO
NCOS 7
SGRP 0
RNPG 0
SCI 0
SSU
XLST
SCPW 1234
SFLT NO
CAC_MFC 0
CLS UNR FBA WTA LPR MTD FNA HTA TDD HFD CRPD
MWA LMPN RMMD AAD IMD XHD IRD NID OLD VCE DRG1
POD SLKD CCSD SWD LND CNDD
CFTD SFA MRD DDV CNID CDCA MSID DAPA BFED RCBD
ICDD CDMD MCTD CLBD AUTU
GPUD DPUD DNDD CFXD ARHD FITD CLTD ASCD
ABDD CFHD FICD NAID BUZZ AGRD MOAD
UDI RCC HBTD AHA IPND DDGA NAMA MIND PRSD NRWD NRCD NROD
DRDD EXR0
USRD ULAD RTDD RBDD RBHD PGND OCBD FLXD FTTC DNDY DNO3 MCBN
FDSD NOVD VOLA VOUD CDMR PRED RECD MCDD T87D SBMD ELMD
MSNV FRA PKCH MWTD DVLD CROD ELCD VMSA
CPND_LANG ENG
RCO 0
HUNT 1822
LHK 0
PUID
UPWD
DANI NO
AST
IAPG 0
AACS NO
ITNA NO
DGRP
MLWU_LANG 0
MLNG ENG
DNDR 0
KEY 00 SCR 58117 0 MARP
01 HOT U 2958117 MARP 0
02
03
04
05
06
07
08
09
10
11
12
13
14
15
16
17 TRN
18 AO6
19 CFW 16
20 RGA
21 PRK
22 RNP
23
24 PRS
25 CHG


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

On the DCH, I see this when calling from UEXT to ip phone:

DCH 100 IMSG SETUP REF 00008181 CH 100 0 0 0 TOD 14:36:10
CALLING #:58117 NUM PLAN: NUM UNKNOWN/UNKNOWN (UNKNOWN)
CALLED #:2958117 NUM PLAN: PRIVATE/ABBREVIATED (CDP)


Thank you

 

[tbonz25]

Do you have incoming and outgoing on the dchannel would like to see both messages as all I see is the incoming message.

 

[nortavaya]

This incoming call DCH:

DCH 100 OMSG SETUP REF 0000019E CH 100 0 0 29 TOD 17:01:00
FEAT :CDS
FEAT :NCID
PROGRESS: ORIG ADDR IS NOT ISDN
CALLING #:2908 NUM PLAN: PRIVATE/ABBREVIATED (CDP)
CALLED #:58117 NUM PLAN: PRIVATE/ABBREVIATED (CDP)

DCH 100 IMSG CALLPROC REF 0000019E CH 100 0 0 29 TOD 17:01:00

DCH 100 IMSG ALERT REF 0000019E CH 100 0 0 29 TOD 17:01:00
PROGRESS: INBAND INFO OR PATTERN IS AVAIL

DCH 100 OMSG DISC REF 0000019E CH 100 0 0 29 TOD 17:01:04
CAUSE :NORMAL CALL CLEARING


Now, I cannot see outgoing DCH message, only busy tone

 

[tbonz25]

Ok so we need to look at the NCOS I see you have it set to 7 on the phone what is the NCOS of your virtual trunks and if this is an outbound call what is the NCOS of your PSTN trunks as well.

 

[nortavaya]

On the IPTI (members of route) I have NCOS 0, but I think it doesn't impact, because I use UNR as CLS on all members and UEXT phone

 

[tbonz25]

I think you still have to have an NCOS high enough on the IPTI trunks to make a call out I would match it up with the PSTN trunks.

 

[nortavaya]

I just configure now all IPTI trunks with the same NCOS 7, but still busy tone and no message output on DCH monitoring
I think the call isn't reaching yet the DCH

 

[tbonz25]

Your phone is registered correct? The other thing is normally you would set the prompt MCCL on the phone to be SIP3 1 to indicate that it is a third party phone and is using 1 client license instead of yes at that prompt.

 

[nortavaya]

Yes, it is registered correclty (the incoming calls are working fine), I can see it when typing command "slgSetShowAll" from SS

Also the phone is already enable SIP3 licence:

MCCL YES
SIPN 0
SIP3 1

 

[tbonz25]

Ok so you can receive an inbound call..when you make an outbound call is it out to the PSTN? Can you dial an internal extension from the SIP phone? If so I would suggest looking at the routing of the external number you are dialing and make sure it is routable. Do you have existing IP phones in place? What is controlling your dialplan for VOIP is it an NRS or Session Manager? Let me know it looks like we may need to dig deeper into this.

 

[nortavaya]

Thank you tbonz25 for your answer

We alreay have internal IP Phone (Unistim), when we try to call them through SIP Phone (third party) we got busy tone

We have NRS, but we didn't configure any of dialing plan, I don't think will use NRS for SIP Line

 

[tbonz25]

Can you call from the Unistim phone to the SIP phone? Did you already have a virtual dchannel built and I am assuming you only have one virtual dchannel correct?

 

[nortavaya]

Yes, From Unistim phone to SIP Phone (3Party) is working fine

As per documentation, I have built the DCH and it is ACTIV

 

[nortavaya]

I have enable PCAP tool on my SS, and when check the SIP messages exchange between SIP Phone and SS, I can see this:

1. Invite sent from SIP Phone to the SS as: sip:1000@sip.exemple:5070;user=phone;transport=tcp
2. Sig Server return: 407 Proxy Authentication Required
3. SIP Phone sends another INVITE with SDP
4. Sig Server return: 403 Forbidden

Any idea ?

Thank you

 

[tbonz25]

Can we verify the NCOS to FRL configuration in ld 87. In LD 87 do a prt and then type is NCTL when it asks for the NCOS just enter 0 7 and verify that they match i.e ncos 0 = frl 0 ncos 1 = frl 1 and so on. Also you said you built the dchannel but just wanted to verify you have only 1 virtual dchannel as all VOIP sipLine or unistim share the same one.

 

[nortavaya]

Yes, I have checked the NCTL, we have NCOS 0 match FRL 0, NCOS 1 match FRL 1 and so on...