sundance-communications.com/forum Forums TDM Business Telephone Systems NEC Telephones & Systems SIP Trunk between NEC SV8100 & Avaya G450

I'm testing SIP trunk between SV8100 & Avaya G450. Avaya G450 is SIP Provider. Successful in Register to Avaya SIP Server and can make call from Avaya G450 to SV8100. But I can not make outgoing call from SV8100 to G450 ( Trunk Group 1 Access code 9 + G450's Extension number)
All commands listed as below:

1. PRG 10-40-01 IP Trunk Availability: Active
2. PRG 10-40-02 SIP Trunk Port: 4
3. PRG 10-28-01 Domain Name
4. PRG 10-28-02 Host Name
5. PRG 10-28-04 User ID
6. PRG 10-28-05 Domain Assignement: Domain Name
7. PRG 10-29-01 Outbound Default Proxy: Active
6 PRG 10-29-03 Default Proxy IP Address
7. PRG 10-29-05 Register Mode: Manual
8. PRG 10-29-11 Registrar Domain Name
9. PRG 10-29-12 Proxy Domain Name
10.PRG 10-29-13 Proxy Host Name
11.PRG 10-30-02 User Name
12.PRG 10-30-03 Password
13.PRG 14-05 - Trunk Group 1 for 4 SIP trunks
14.PRG 22-02 - Incoming Call Trunk Setup: DID
15.PRG 22-11 : DDI Translation Table

Anybody have idea about the missing configuration for the SIP trunking?

You need to "enblock" the out going number. Try appending a "#" I have seen this work on some systems. To work properly you will need to set up an FROUTE table for extension numbering so that you can dial 3 or 4 digits.

try 22-19. Many times you need an outbound CID to get through.

You may need to look at 10-23 and 10-40

Thanks for all your advice.I already tried all but still not work
@Coral Tech: I dial 9 + telephone number + # and also the F-route with limited digit dialing
@helpifican: Do you mean 21-19 IP Trunk (SIP) Calling Party Number Setup for Extensions? I put the SV8100 extension number in, but couldn't make call also
@dans: I choose the Registration Mode. Should I put my SV8100 IP address in the 10-23 for the Interconnection setup?

What happens when you attempt the call?

If you haven't already, you might try putting the SIP trunk you expect to dial out on onto a line key. When you dial the number do you see it access the SIP trunk on that line key?

The call is released if I use F-route / 9+extension + # to make call. Busy tone (release tone) is returned back.

I access trunk line 001 to make outgoing call. This trunk line is OK for incoming call.

I use DIM log to get log file for unsuccessful call but no idea for this log:

19:41:43 >>
19:41:43 >><<< cdialsta.cpp(311) Originator dial buffer already opened >>>
regId = 0
CALLED cnetCheckARSDigit() lport=401
ITR_NULL_P_STA: DES_ANSWER_FLAG CLEAR!
ITR_NULL_P_STA: TM_DES_ANSWER_WAIT_L SET!
** CLR_ISDN_FLAG IS CALLED **
** LPORT_W = dc01H **
** CLR_ISDN_FLAG IS CALLED **
** LPORT_W = dc01H **
PK06E10162 PRT CPN num_type == 0, num_plan == 0
create_called_no: size_b 0040H
Call cntrol Recived CAPS

[CC_task data]
Logi_port:0xdc01
len:0x2d
msg_gen:0xa1
primitive:0x04
Line:0x01
Ces:0x01
sapi:0x00
0x08 0x02 0x00 0x04 0x05 0x04 0x03 0x80
0x90 0xa3 0x18 0x03 0xa9 0x83 0x81 0x6c
0x06 0x00 0x81 0x33 0x39 0x30 0x34 0x70
0x05 0x81 0x33 0x34 0x30 0x32 0x7c 0x03
0x80 0x90 0xa3 0x7d 0x02 0x91 0x81 0x00
0x00 0x00 0x00 0x00 0x00
Target System ID is 0
msgPrim :0x04
anaRes.a.crInfo.crSide :0x01
anaRes.a.crInfo.crValue :0x04
š[ipCcRecvCAPS] regId = 0 š
š[ipCcRecvCAPS] p2pChkFlag2, s_logicalPort2 Initialize(0xff)
[ipCcRecvCAPS]CHECK FROM CAPS callInfo_p=(4baa150)
Logical_port:0xdc01
DSP Port 0x101
>>ipCcSipP2PSetting[dc01] invalid spid
RTP PORT:0x2726
RTP Adder:0xc0a81083
Logical_port:0xdc01
IP Adder 0xffffffff
ipCcSipSE() call!!
[ipCcSipSE] : PRIMITIVE (4) Recieved from UPPER_TASK
Create New Call date 01/01/0108 time: 19:41:49
suConnId(15) crValue(4) crSide(1) ccSipCb(4baa21c) size(5300)
>>ipCcSipP2PSetting[dc01] OFF
>>ipCcSipP2PSetting[dc01] OFF
[OGGW] [SETUP->INVITE] state(0) suConnId(15) spConnId(0) crValue(4)
/--- INVITE format information ---/
input format length = 512
made format length = 118
format =
INVITE sip:%s SIP/2.0

From:"%s"

To:

Contact:

Content-Type:application/sdp







[SGW] State(0)->(3) suConnId(15) spConnId(0) crValue(4)
Function Call_cntrol_req
>>>[ipCcRecvSIP]Incoming SIP mesage spid=2 ,cLeg->ifEvent = 21
>>ipCcSipP2PSetting[0] OFF
New CLegId Detected(80000000)
[ICGW] [DISC_IND<-RELIND] state(3) EV(21) EVTYPE(21) MSGTYPE(3) suConnId(15) spConnId(0) crValue(4)
[SGW] State(3)->(c) suConnId(15) spConnId(0) crValue(4)
[ipCcRecvSIP]ipCcFindCr callNum_p=(4baa0c8) cinfo.callNum.crValue=(4) side=(1)
[ipccActivateOtherSE] ipCcCloseRsrc call !!
[ipCcSipSE] : PRIMITIVE (9) Recieved from SIP_TASK
Function Call_cntrol_req OK
T305 TIMER CLEAR !!!
multi_call_control_pro Before-Status:0

multi_call_control_pro After-Status:0

08 02 80 90
GET_CAUSE: Returned value is 16
08 02 80 90
GET_CAUSE: Returned value is 16
Call cntrol Recived CAPS

[CC_task data]
Logi_port:0xdc01
len:0x0f
msg_gen:0xa1
primitive:0x0c
Line:0x01
Ces:0x01
sapi:0x00
0x08 0x02 0x00 0x04 0x4d 0x08 0x02 0x80
0x90 0xa3 0x18 0x03 0xa9 0x83 0x81
Target System ID is 0
msgPrim :0x0c
anaRes.a.crInfo.crSide :0x01
anaRes.a.crInfo.crValue :0x04
[ipCcRecvCAPS]CHECK FROM CAPS callInfo_p=(4baa150)
Logical_port:0xdc01
IP Adder 0xffffffff
ipCcSipSE() call!!
[ipCcTimerSet] !START! (PR_SIP_TIMER_DISC)timerparam->timerval = 60
[ipCcTimerSet] Disc Timer SET! index = 0
[ipCcSipSE] : PRIMITIVE (12) Recieved from UPPER_TASK
Calling party number IE is not available
>>ipCcSipP2PSetting[dc01] OFF
>>ipCcSipP2PSetting[dc01] OFF
[OGGW] [REL->OKor486] state(c) suConnId(15) spConnId(0) crValue(4)
[SGW] State(c)->(0) suConnId(15) spConnId(0) crValue(4)
[ipCcTimerSet] (PR_SIP_TIMER_DISC) !STOP! timerparam->info = 0, timerparam->timerval = 0
[ipCcTimerSet] Disc Timer Stop! index = 0
Delete Call suConnId(15) crValue(4) crSide(1) ccSipCb(4baa21c)
Number of Active call(0)
[ipCcSipSE]:STAT_SIP_NULL -> clean
[CLR rsrcList_p] callInfo_p->rsrcList_p sets NULL!!
ipCcTrRecfgSend() called!!!
Function Call_cntrol_req
Function Call_cntrol_req OK
[rtp_ipinfo]>>>Obtaininig Semaphore is SUCCESS!!.
[rtp_ipinfo]>>>Release Semaphore is SUCCESS!!.
***> Call Information is released. (0x04baa150) <***
T305 TIMER CLEAR !!!
multi_call_control_pro Before-Status:0

multi_call_control_pro After-Status:0

TRK TARGET_DIAL=00000000
hunt_que_del_allstg des_w[dc01]
== Initialize rgt_port_data memory for port:dc01
== rgt_port_data_read(dc01) was 0000
== rgt_des_data_read(dc01) was 0000
ipCcTrRecfgSend() called!!!
Function Call_cntrol_req
Function Call_cntrol_req OK
TRK TARGET_DIAL=00000000
cmain.cpp(3962) Clear RGT Port : Port = 401 Rgt Port = 0
cmain.cpp(3964) Clear RGT des Port : Port = 401 Rgt des Port = 0

I haven't done this before, so my advice maybe not right.
since you could make phone calls from avaya to sv8100, there is nothing wrong with PRG 22-XX.

IF you use F-route, you don't need Access code 9 to get the sip trunks, you could dial the extension numbers directly. I mean they are different things.

IF you could get sip trunk through dialing 9, you could also check configurations on avaya system.

Crazyfrog:
Thanks for your advice.
Yah, actually, I tried make call using 2 ways - and both not successful.

1. F-route, I just dial the Avaya extension only.
2. Trunk access code 9 + Avaya extension + #

I searched on internet, there are some suggestion that:

- PRG 20-08-13 - ISDN CLIP: activate
- PRG 21-19 - IP Trunk (SIP) Calling Party Number Setup for Extension

- PRG 84-13-32 DTMF Relay Mode: RFC2833
But it didn't get effect also.

I changed some configuration and the outgoing call is working now:

1. PRG 10-28-01 Domain Name: to IP address
2. PRG 10-28-02 Host Name: to IP address
3. PRG 10-28-05 Domain Assignement: IP address

Thank you so much for all you guys for reading and supporting for on this topic.

If you need the screenshot of setting, please PM me.

Dear Sir ,

Please send me the programming screen shoot for both side and send them to mail [email protected]

Hi Mohammad,

Already sent email to you. Hope it help.

Yes , I received it many thanks ...

Dear sir,

Please send me the programming screen shoot for both side
I'm on the same problem

Nec SL1000 ---sip trunk Asterisk

can make phone calls from asterisk ---> NEC (extensions )

From Nec ---> Asterisk .....not working

Hi Amorphys,

Send me your email address, please

Thnks i received the files....let's see....now

Sir I am Connecting SIP Trunk between SV8100 and Panasonic NS 500.In Dubai we are using Panasonic NS500 System. through which dubai to india dialing is working fine. but for india to dubai dialing is not working. the engage dial tone is coming.

We have setup VPN through Fortigate firewall in which right now we have given access for all ports in india & dubai firewall. dubai is also using same fortigate firewall in which all ports are open.
Dubai user are using 900 series .
India users are using 800 series.
We were using H.323 on both side. From Nec we came to know that H.323 connects only to NEC system. So now we are trying to connect it on SIP.

I just want to know whether NEC system can act as a SIP Server.

Hi Afroz,
I got no idea if SV8100 can act as a SIP Server. In SV8100, I only found 2 options: Registration mode & Peer to Peer.

Thanks November,

If Peer to Peer is possible than what would be steps, Can you help me in this.

Your support is highly appreciated.

Hi Afroz,
I have doc for peer to peer mode. However, I just tested between NEC products. If you still need it, send me your email address

Dear November

could you please send me again the screen shots i lost your email

[email protected]

dear all, i want to connect via a sip trunk between NEC SV8100 and AVAYA G450.
i need a snap shut of the configuration.

thank you

Dear November I have the same problem can you send the files please [email protected]

Here's a friendly tip: Before you reply to a post, CHECK THE DATE! My calendar says it 2025. Replying to a post or reaching out to a member who has not posted or been seen in 5-10 years is going nowhere.

It's also hard to believe you have the "exact same problem" on a phone system that has been discontinued for 6 years. Where were you 6 years ago?

Anyway, a while back ago, we had a discussion about rules, but decided to just keep the board open to older posts. I checked 'November's links and they are broken. It probably means they are no longer around. Since our post traffic is so small, we can see and/or respond to any post. I try to help whenever possible, but I am not your library and software supply company. You might try making a new post, listing what you have done and why you're trying to interface an SL-1100 or 8100 family to asterisk and what you have tried to solve your problem. Then someone might have answers.

Just my two cents....

Carl