- This topic has 7 replies, 1 voice, and was last updated 20 years, 5 months ago by Mark.
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25th June 2004 at 17:28 #27497HamyGuest
Here is the scenario:
A800 Analog
Calls originated from PBX to IP
If the call fails over IP it goes to PSTN
I have 2 questions:
1- Is it possible to only reroute calls that fail with a specific release cause to PSTN? It seems A800 reroute all the calls (even wrong numbers) to PSTN. Can I prevent it?
2- When the call reroutes to PSTN, A800 considers the call connected while the call is in progress and it connects after 10 or 20 seconds. The CDR logs the call duration from the moment the call is routed to PSTN and this means an extra charge for the customer. Is there a way to stop this and let the tenor only start the calculation of the call duration when the call is actually connected?
Any help would be really appreciated.
Regards28th June 2004 at 21:07 #27498MarkGuestHamy,
1. The Quintum Tenor should only re-route to the PSTN when IP calls fail for the following cause codes 3, 31, 34, 38, 41-45, & 47. It will also re-route on code 17 if partial trunk group is enabled.
2. This is an issue where you have not setup the answer supervision correctly. In the PSTN trunk group , set supervision to 2 (answer) or 3 (answer & disconnect), set the answerdelay to 120 and make sure progtone is set to 0. For more information on this review the answer supervision document on Quintum’s web site at;
http://www.quintum.com/support/1G/kb/telco/Answer_Supervision.pdf29th June 2004 at 08:14 #27499HamyGuestDear Mark,
Thank you for your answer. I have set it like this (PSTN 1):
1- cassig 1
2- super 2
3- answer 120
4- progtone 0
and DSP:
1- asop 3
It seems I still have the problem. Tenor waits 120 seconds and then starts billing. Anything I am doing wrong?
System Software Version: P4-2-20-38(LEC) (1733185/0x81C5)
Boot Software Version: P4-1-3 (180592/0xE814)
Database Version: 2.08 09-13-2000 (277900)30th June 2004 at 10:34 #27500HamyGuestAnybody? Please!
1st July 2004 at 15:43 #27501MarkGuestTry setting the asoptions to 0.
2nd July 2004 at 16:32 #27502HamyGuestThank you for your answer. Have done that. No success 🙁 .
Here comes the configuration:
———————-
Unit
—-
Unit: 1
IP Address = 192.168.254.2
External IP Address = ***************
Name = ctel1Online = 1
Relay ResetTime = 240
Relay Reset Number = 2TcpKeepAlive = Disabled(0)
System
——
Country Value = 6
Country Code = 32
Area Code = 2
Minimum DN = 7
Maximum DN = 7
International Prefix:
1: 00
Long Distance Prefix = 0
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 = 0.0.0.0IP 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 = 16IP 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:
Cdr Format: 0Ring 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: UnknownDisc 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 nprefixSystem LAN
———-
Subnet Mask = 255.255.255.240
Default Gateway = 192.168.254.1PSTN Trunk Group
—————-
PSTN Trunk Group: 1
Name = pstn1
Pass Through = yes(1)
PT Trunk ID = 0
Provide Call Progress Tone = no(0)
Busyout = no(0)
Hunt Algorithm = ascending(0)
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 = 1
LAM: Index Pattern Replacement NumberType
Cas Signaling Type = loop start fwd disconnect(6)
Cas Orientation = user(0)
Dial Tone Detect = yes(1)
Dial Delay Timeout = 1000
Answer Delay Timeout = 120
Flash-Hook Signaling = no(0)
Supervision = answer and disconnect(3)
Caller Id Detection = no(0)
dtmf-ontime = 100
dtmf-offtime = 100
Channel:
unit# 1 line# 2: 1PBX Trunk Group
—————
PBX Trunk Group: 1
Name = PbxPassThrough1
Pass Through = yes(1)
PT Trunk ID = 0
Provide Call Progress Tone = no(0)
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 = 16
Private Number of Digits = 4
Public Hunt Ldn’s:
Private Hunt Ldn’s:
BDN: Index Bdn
1 02***********
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: 1,2,3,4,5,6,7,8IP 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-GatewayLine
—-
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 = EnabledBandwidth 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
1 192.168.254.2 255.255.255.255Barred Endpoints
IP Mask
No Barred Endpoints ConfiguredGatekeeper System
—————–
Zone Name =
Border Element IP Address(prim) = 192.168.254.2
Border Element IP Address(sec) = 0.0.0.0
Discovery IP Address = 192.168.254.2
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
Static Route #1
RouteName =
Gkmode = Destination is a Gateway (0)
CallSignalAddress = **********************:1720
1:1* Public LDN priority(0)
2:2* Public LDN priority(0)
3:3* Public LDN priority(0)
4:4* Public LDN priority(0)
5:5* Public LDN priority(0)
6:6* Public LDN priority(0)
7:7* Public LDN priority(0)
8:8* Public LDN priority(0)
9:9* Public LDN priority(0)DSP
—
Voice Coding algorithm = 81
Voice Information Field size = 192 bits
Silence Suppression = Enable(1)
Minimum Jitter buffer = 60 msec
Maximum Jitter buffer = 300 msec
Receive Gain (PCM -> IP) = -2 dB
Transmit Gain (IP -> PCM) = -4 dB
Digit Relay = 0
Fax Relay Type = 1
T.38 Fax Low Speed Data Redundancy = 0
T.38 Fax High Speed Data Redundancy = 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 = 0AutoSwitch
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Auto Threshold = 50H323 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 = No(0)
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 1813host s 0.0.0.0
authenticationport s 1812
accountingport s 1813retry = 3
timeout = 5
accountingtype = 0
billingvendor 0
sharedsecretIVR
——Primary File Server: IP Address = 0.0.0.0
Secondary File Server: IP Address = 0.0.0.0
timeout: 5Enabled Languages: None
CID Translation Table
——
Caller ID Translation Table
Index Pattern ReplacementRadius Endpoint
—————host p 0.0.0.0
authenticationport p 1812
accountingport p 1813host s 0.0.0.0
authenticationport s 1812
accountingport s 1813retry = 3
timeout = 5
idtype = 0
passwordtype = 0
sharedsecret
————————–
When I dial 02*********** tenor waits 120 seconds and then starts billing. Anything I can do about it?
Best regards7th July 2004 at 11:21 #27503HamyGuestAnyone? I really nedd to get this to work. Thank you for answering.
7th July 2004 at 15:27 #27504MarkGuestTry setting the cassig to 1. I do not think this will change the answer problem, but it is a conflict as you cannot have forward disconnect when supervision is set to both answer and disconnect.
After this, without going through logs and testing, it would be hard to determine what is happening.
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