tenor A800

Saroj,

there could be many reasons for this and without any further troubleshooting it would be difficult to say what is the cause. If, as soon as call hits the gateway you get fast busy, but then the cause code is 16, I find that somewhat strange as typically in this case you may get a cause 41, or 34 or 47.

This would require additional trouble shooting, take logs, look at config and such.

If you would like assistance on this, please contact me at mike_voip@hotmail.com and we can discuss my services.

Mike M
mike_voip@hotmail.com

Mizanur,

There are many things to consider for doing term.

If you would like assistance in configuration, you may contact me directly at mike_voip@hotmail.com and we can discuss my services.

MikeM
mike_voip@hotmail.com

Can anyone tell me how can i eliminate de Border Element.

Hello Mike,

Thanks The Tenor A800 is now working and the reason for not being able to configure it was the Serial port mismatch due to the upgraded versions of available computers. I used a USB to serial convertor and it worked.

Thanks for the assistance and hope we shall chat again on MSN soon.

With Kindest Regards
Riaz Malik

Saroj,

If you are simply looking to connect a phone to the A800 and make a call out over IP, you need only configure a static route with the destination IP address and add a * as the DN so this way any calls you dial will go to that IP address. You may want to make sure that you have the country set to 1, countrycode removed, areacode removed, ldprefix removed and intlpref 1 removed. These parameters are all found under the config sys prompt.

If you need further assistance you can contact me at mike_voip@hotmail.com to discuss my services.

thanks
Mike M
mike_voip@hotmail.com

Here is my configuration.

Unit
—-
Unit: 1
IP Address = xxx.xxx.xxx.xxx
External IP Address = 0.0.0.0
Name = nepalroc

Online = 1
Relay ResetTime = 240
Relay Reset Number = 2

TcpKeepAlive = Disabled(0)

System
——
Country Value = 3
Country Code =
Area Code =
Minimum DN = 7
Maximum DN = 20
International Prefix:
Long Distance Prefix =
Carrier Selection Prefix:
Intercom Used = no(0)
Private DN Used = no(0)
Interdigit Timeout = 4 sec.

Contact =
Location =
IP Address : of Snmp Trap Server 1 = 0.0.0.0
IP Address : of Snmp Trap Server 2 = 0.0.0.0
IP Address : of Snmp Trap Server 3 = 0.0.0.0

IP Address : Port # of Syslog Server 1 = 0.0.0.0 : 514
IP Address : Port # of Syslog Server 2 = 0.0.0.0 : 514
IP Address : Port # of Syslog Server 3 = 0.0.0.0 : 514
Syslog Facility = 16

IP Address : Port # of Cdr Server 1 = 0.0.0.0 : 0
IP Address : Port # of Cdr Server 2 = 0.0.0.0 : 0
Cdr Password: key
Cdr Format: 0

Ring Frequency = 20 Hz(0)
PSTN Ring Sensitivity = Normal(0)

Primary Time Server: IP Address = 0.0.0.0
Secondary Time Server: IP Address = 0.0.0.0
UTC Offset: Unknown

Disc Tone Frequency: 480 Hz (min) : 620 Hz (max)
Disconnect On/Off Time: 250 mSec(on) : 250 mSec(off)

Call Indication Tone = None(0)
Disable GUI = no(0)

Dialplan
——–
User Programmable DP = No
Dialplan table:
index:Pattern DpType min max nprefix

System LAN
———-
Subnet Mask = 255.255.255.0
Default Gateway = xxx.xxx.xxx.xxx

PSTN Trunk Group
—————-
PSTN Trunk Group: 1
Name = PstnPassThrough1
Pass Through = yes(1)
PT Trunk ID = 0
Provide Call Progress Tone = no(0)
Busyout = no(0)
Hunt Algorithm = ascending-round-robin(2)
Modem Calls = No(0)
Direction = outgoing(1)
DN Used = public
End Of Dial = yes(1)
End Of Dial Digit = #
Add End of Dial Digit = no(0)
Ivr Type = None
CID = From Interface(0)
External Routing Request = no(0)
Auto Switch Enable = no(0)
Forced IP Routing # = none
Trunk ID(Account Code) = none
Trunk ID Delivery = none
2 Stage Dial = No
Translate Inbound Caller ID = no(0)
Relay Caller ID = yes(1)
IP Extension = no(0)
Maximum LAM Calls Allowed = 8
LAM: Index Pattern Replacement NumberType
Cas Signaling Type = loop start rev battery(5)
Cas Orientation = user(0)
Dial Tone Detect = yes(1)
Dial Delay Timeout = 1000
Answer Delay Timeout = 0
Flash-Hook Signaling = no(0)
Supervision = none(0)
Caller Id Detection = no(0)
dtmf-ontime = 100
dtmf-offtime = 100
Channel:
unit# 1 line# 2: 1,2,3,4,5,6,7,8

PBX Trunk Group
—————
PBX Trunk Group: 1
Name = PbxPassThrough1
Pass Through = yes(1)
PT Trunk ID = 0
Provide Call Progress Tone = yes(1)
Multipath = yes(1)
Hunt Algorithm = ascending(0)
Modem Calls = No(0)
Direction = both(2)
DN Used = public
End Of Dial = yes(1)
End Of Dial Digit = #
Add End of Dial Digit = no(0)
Ivr Type = None
Partial TG = no(0)
CID = Trunk ID(1)
External Routing Request = no(0)
Auto Switch Enable = no(0)
Forced IP Routing # = none
Trunk ID(Account Code) = none
Trunk ID Delivery = none
2 Stage Dial = No
Translate Inbound Caller ID = no(0)
Relay Caller ID = yes(1)
IP Extension = no(0)
Public Number of Digits = 7
Private Number of Digits = 4
Public Hunt Ldn’s:
Private Hunt Ldn’s:
BDN: Index Bdn
Cas Signaling Type = loop start(1)
Cas Orientation = net(2)
Flash-Hook Signaling = yes(1)
Flash-Hook Min = 200
Flash-Hook Max = 700
Supervision = none(0)
Caller Id Generation = no(0)
dtmf-ontime = 100
dtmf-offtime = 100
Channel:
unit# 1 line# 1: 1,2,3,4,5,6,7,8

IP Trunk Group
—————
Incoming IP call delete digits = 0
Incoming IP call prefix =
Outgoing IP call delete digits = 0
Outgoing IP call prefix =
Prefix Trunk ID = no(0)
Default Trunk = No
External Routing Request = no(0)
Display Information ID = Tenor-Gateway

Line
—-
Line: 1
Law = uLaw(0)
Rx Gain = -4dB
Tx Gain = -2dB
Line: 2
Law = uLaw(0)
Rx Gain = 0dB
Tx Gain = 0dB
Guard Time = 0mS
DAA Start Up = Enabled

Bandwidth Management
——————–
Time of Day Maximum Bandwidth:
Day = 0(Sunday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *
Day = 1(Monday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *
Day = 2(Tuesday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *
Day = 3(Wednesday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *
Day = 4(Thursday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *
Day = 5(Friday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *
Day = 6(Saturday)
Hour = 00 * * * * * *
Hour = 06 * * * * * *
Hour = 12 * * * * * *
Hour = 18 * * * * * *

Gatekeeper Administration
————————-

Endpoint Authorization Type = 0 (None)

Allowed Endpoints
IP Mask
No Allowed Endpoints Configured

Barred Endpoints
IP Mask
No Barred Endpoints Configured

Gatekeeper System
—————–
Zone Name = npRoc
Border Element IP Address(prim) = xxx.xxx.xxx.xxx
Border Element IP Address(sec) = 0.0.0.0
Discovery IP Address = 0.0.0.0
Gatekeeper Password =
LRQ returns all candidates(0)
Maximum LRQ Hops = 0
WAN Call Limit = 0 (disabled)
LCF/LRJ V3plus = 1
Gatekeeper Option Flags:
Use IP Header Address = no(0)
Ridgeway ARQ = no(0)

Border Element
—————–
Static Routing
No Static Routes configured

DSP
—
Voice Coding algorithm = 68
Voice Information Field size = 160 bits
Silence Suppression = Enable(1)
Minimum Jitter buffer = 60 msec
Maximum Jitter buffer = 300 msec
Receive Gain (PCM -> IP) = 0 dB
Transmit Gain (IP -> PCM) = 0 dB
Digit Relay = 1
Fax Relay Type = 0
Fax Maximum Rate = 144
Fax Playout FIFO nominal delay = 600
Fax Modem Coding = 0
Fax Modem Voice Information Field size = 0 bits
Idle Time = 0
Answer Supervision Options = 0
Disconnect Supervision Option = 0
Answer Supervision Delay = 0 (disabled)

AutoSwitch
———-
Auto Threshold = 50

H323 Gateway
—————–
Primary GK Address = 0.0.0.0
Primary Auto Discovery = 1
Secondary GK Address = 0.0.0.0
Secondary Auto Discovery = 0
H.323 ID =
Register DN = Register as GW Prefixes(1)
Ignore Bandwidth in ACF = no(0)
Default H245 Tunneling = yes(1)
Round Trip Delay = 0
One Stage Dialing = 0
RRQ Interval = 0
H323 Interop Flags:
H225 GW Protocol = h323(0)
Do BRQ = 0
SlaveSessionId0 = 0
AllowFastStartOnly = Yes(1)
RRQv3plus= No(0)
ProgressInd Alert= No(0)
StartH245Flag= No(0)
Automatic Ext IP Update= No(0)
RTP Verification= No(0)

Do Lightweight RRQ = no(0)

Radius User
———–

host p 0.0.0.0
authenticationport p 1812
accountingport p 1813

host s 0.0.0.0
authenticationport s 1812
accountingport s 1813

retry = 3
timeout = 5
accountingtype = 0
billingvendor 0
sharedsecret

IVR
——

Primary File Server: IP Address = 0.0.0.0
Secondary File Server: IP Address = 0.0.0.0
timeout: 5

Enabled Languages: None

CID Translation Table
——
Caller ID Translation Table
Index Pattern Replacement

Radius Endpoint
—————

host p 0.0.0.0
authenticationport p 1812
accountingport p 1813

host s 0.0.0.0
authenticationport s 1812
accountingport s 1813

retry = 3
timeout = 5
idtype = 0
passwordtype = 0
sharedsecret

Also,

Please do not copy your entire config to these forums as they are long enough without everyone having to read through your config.

Again, as stated, check out the docs on quintum’s web site because in the pstntg area you definitely do not have the correct settings for answer and disconnect supervision. I would try setting signaling to just loop start, supervision to 3 (both answer and disconnect) and answer delay to 120.

Mike M
mike_voip@hotmail.com