SPA-3102 Gateway PSTN Outbound Dialling Problem

[invisage]

SPN-3102 Gateway PSTN Outbound Dialling Problem with Non Commercial Versions of 3CX.

Fault discription. I can receive incoming pstn phone calls through the 3102 without any problem but cannot place outbound phone calls.
I have setup the 3102 as directed here http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE
Installed and configured a voip line. All voip outgoing calls tested and worked Ok.

Any help would be great appreciated!

See attached Jpg files of my configuration setup pages. Phone system based in Australia.

Things I have tried:-
Have setup 3102 as just voip gateway as directions above link.
Have setup 3102 as both voip gateway and with ata functionality as directed above link.
Created test as above with clean xp pro install with 3cx v5 & v6.
Done a dance on one 3102 then purchased another.
:twisted:

 

[Cjay]

It looks to me as if you may be trying to route calls with 3 digits to your Landline 1 (3102?). Does 3 digits conflict with your internal 3cx extension number plan?

Why don't you try removing the 3 digits and simply force all calls out via the 3102?
For example, anything beginning with 0 goes to the 3102:
In manage outbound calls create a rule:
Name = Anything you like!
Apply to extensions = Leave blank (so will apply to outgoing calls from all extns)
Number Prefix = 0
Number Length = Leave blank (so length of dialled number is irrelevant to routing)
Route 1 = Name of your 3102 gateway

Then having completed this go into manage outbound rules, click on the name of the new rule and check strip/prepend. Based on the above example with dialled numbers beginning with 0, If you set strip=0 and prepend = (blank), then if you dial 0123456789 on a 3cx extension then this is exactly what goes out to the 3102.

Outbound rules are explained in the manual!

Chris

 

[invisage]

Thanks Chris, tried your idea and created new call rule. (See new attachment) dialed outside number with a prefix of 0, unfortunately no go. Just get the same beep beep beep etc tone again. I do see the extension light on the 3CX consol of the phone extension I used to dial go green for a moment. Nothing on the 3102 line light. But I think that has always been the case though. It is strange all the lights & bells etc work with incoming calls.

My gut feeling is it is a configuration issue with the 3102 and perhaps our phone system infrastructure.

Regards Geoff

 

[Cjay]

What does the 3cx log show for a failed outgoing call?

 

[invisage]

Just tried to phone outside number 048771123 again and got this from the call log. is this the log you are refering to Chris.


2008-07-23 20:02:38 110 048771123 No Answer

 

[invisage]

Oops Sorry just relised what report you needed!
Its interesting looks like the outbount is trying to use the voip line. I am using the 0 prefix before dialing. What the?
Could this mixup be because both the 3102 & voip lines are using the same port 5060?

 

[invisage]

Ok just disabled outgoing calls on the VoIP line and reset call rules al to go through the 3102 and now get this result.
also i changed PSTN Lines > Sip port to 5060 as it was set to 5061 in the 3102 config setup page.

 

[invisage]

Noticed others in the forum have had similar problems and cannot dial out on there 3102 also. Is it correct that to resolve the fault you need the commercial version of 3CX?

 

[Anonymous]

invisage,

I notice from your screenshots that "Auth ID" is blank. This should be enabled.

Also, please set "Make calls without Reg" to "No".

The issue here has no relation to commercial/non-commercial versions.

Let us know of your mileage.

 

[Henk]

PSTN Line:

SubscriberInformation:

USer ID and Auth ID make them both the same.

FXO Timer Values:
PSTN answer delay: 3
PSTN Ring Thru Delay: 1.5

Try that,

H.

 

[techworx]

i also have the same problem ,tryed everything , online guide, henks manual, all tips i can find on site.

result,incoming no problem , outgoing goes imeddiately engaged.

i have posted my server and pstn page below

using linksys 942

maybe i should trow it away and get grandstreams, cannot get patton in thailand

Time Function Message
17:39:47.703 Call::Terminate [CM503008]: Call(5): Call is terminated
17:39:47.656 Call::RouteFailed [CM503015]: Call(5): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
17:39:47.656 CallLeg:nFailure [CM503003]: Call(5): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:192.168.2.114:5060
17:39:47.609 CallCtrl:nSelectRouteReq [CM503004]: Call(5): Calling: PSTNline:0844439863@(Ln.10000@pstn 1)@[Dev:sip:[email protected]:5061]
17:39:47.593 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
17:39:47.593 Extension:rintEndpointInfo [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-5.1.15(a)] Transport: [sip:192.168.2.10:5060]
17:39:47.546 CallCtrl:nIncomingCall [CM503001]: Call(5): Incoming call from Ext.100 to [sip:[email protected]]



User Login basic | advanced



Product Information
Product Name: SPA-3102 Serial Number: FM600H210525
Software Version: 5.1.7(GW) Hardware Version: 1.4.5(a)
MAC Address: 000E08CE345C Client Certificate: Installed
Customization: Open

System Status
Current Time: 7/28/2003 17:40:01 Elapsed Time: 00:24:40
RTP Packets Sent: 388 RTP Bytes Sent: 93120
RTP Packets Recv: 385 RTP Bytes Recv: 92400
SIP Messages Sent: 8 SIP Bytes Sent: 5128
SIP Messages Recv: 10 SIP Bytes Recv: 4575
External IP:

Line 1 Status
Hook State: On Registration State: Not Registered
Last Registration At: Next Registration In:
Message Waiting: No Call Back Active: No
Last Called Number: Last Caller Number:
Mapped SIP Port: Yahoo!:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:
Call 1 Media Loopback: Call 2 Media Loopback:

PSTN Line Status
Hook State: On Line Voltage: 48 (V)
Loop Current: 0.0 (mA) Registration State: Registered
Last Registration At: 7/28/2003 17:15:32 Next Registration In: 315 s
Last Called VoIP Number: 10000 Last Called PSTN Number:
Last VoIP Caller: Last PSTN Caller: ,
Last PSTN Disconnect Reason: VoIP Call Ended PSTN Activity Timer: 30000 (ms)
Mapped SIP Port: Call Type:
Yahoo!:
VoIP State: Idle PSTN State: Idle
VoIP Tone: PSTN Tone:
VoIP Peer Name: PSTN Peer Name:
VoIP Peer Number: PSTN Peer Number:
VoIP Call Encoder: VoIP Call Decoder:
VoIP Call FAX: VoIP Call Remote Hold:
VoIP Call Duration: VoIP Call Packets Sent:
VoIP Call Packets Recv: VoIP Call Bytes Sent:
VoIP Call Bytes Recv: VoIP Call Decode Latency:
VoIP Call Jitter: VoIP Call Round Trip Delay:
VoIP Call Packets Lost: VoIP Call Packet Error:
VoIP Call Mapped RTP Port:


System Configuration
Restricted Access Domains:
Enable Web Admin Access: yesno Admin Passwd:
User Password:

Miscellaneous Settings
Syslog Server: Debug Server:
Debug Level: 0123


SIP Parameters
Max Forward: Max Redirection:
Max Auth: SIP User Agent Name:
SIP Server Name: SIP Reg User Agent Name:
SIP Accept Language: DTMF Relay MIME Type:
Hook Flash MIME Type: Remove Last Reg: yesno
Use Compact Header: yesno Escape Display Name: yesno
RFC 2543 Call Hold: yesno Mark All AVT Packets: yesno
SIP TCP Port Min: SIP TCP Port Max:

SIP Timer Values (sec)
SIP T1: SIP T2:
SIP T4: SIP Timer B:
SIP Timer F: SIP Timer H:
SIP Timer D: SIP Timer J:
INVITE Expires: ReINVITE Expires:
Reg Min Expires: Reg Max Expires:
Reg Retry Intvl: Reg Retry Long Intvl:
Reg Retry Random Delay: Reg Retry Long Random Delay:
Reg Retry Intvl Cap:

Response Status Code Handling
SIT1 RSC: SIT2 RSC:
SIT3 RSC: SIT4 RSC:
Try Backup RSC: Retry Reg RSC:

RTP Parameters
RTP Port Min: RTP Port Max:
RTP Packet Size: Max RTP ICMP Err:
RTCP Tx Interval: No UDP Checksum: yesno
Stats In BYE: yesno

SDP Payload Types
NSE Dynamic Payload: AVT Dynamic Payload:
INFOREQ Dynamic Payload: G726r16 Dynamic Payload:
G726r24 Dynamic Payload: G726r32 Dynamic Payload:
G726r40 Dynamic Payload: G729b Dynamic Payload:
EncapRTP Dynamic Payload: RTP-Start-Loopback Dynamic Payload:
RTP-Start-Loopback Codec: G711uG711aG726-16G726-24G726-32G726-40G729aG723 NSE Codec Name:
AVT Codec Name: G711u Codec Name:
G711a Codec Name: G726r16 Codec Name:
G726r24 Codec Name: G726r32 Codec Name:
G726r40 Codec Name: G729a Codec Name:
G729b Codec Name: G723 Codec Name:
EncapRTP Codec Name:

NAT Support Parameters
Handle VIA received: yesno Handle VIA rport: yesno
Insert VIA received: yesno Insert VIA rport: yesno
Substitute VIA Addr: yesno Send Resp To Src Port: yesno
STUN Enable: yesno STUN Test Enable: yesno
STUN Server: TURN Server:
Auth Server: EXT IP:
EXT RTP Port Min: NAT Keep Alive Intvl:


Configuration Profile
Provision Enable: yesno Resync On Reset: yesno
Resync Random Delay: Resync Periodic:
Resync Error Retry Delay: Forced Resync Delay:
Resync From SIP: yesno Resync After Upgrade Attempt: yesno
Resync Trigger 1:
Resync Trigger 2:
Resync Fails On FNF: yesno
Profile Rule:
Profile Rule B:
Profile Rule C:
Profile Rule D:
Log Resync Request Msg:
Log Resync Success Msg:
Log Resync Failure Msg:
Report Rule:

Firmware Upgrade
Upgrade Enable: yesno Upgrade Error Retry Delay:
Downgrade Rev Limit:
Upgrade Rule:
Log Upgrade Request Msg:
Log Upgrade Success Msg:
Log Upgrade Failure Msg:
License Keys:

General Purpose Parameters
GPP A:
GPP B:
GPP C:
GPP D:
GPP E:
GPP F:
GPP G:
GPP H:
GPP I:
GPP J:
GPP K:
GPP L:
GPP M:
GPP N:
GPP O:
GPP P:


Call Progress Tones
Dial Tone:
Second Dial Tone:
Outside Dial Tone:
Prompt Tone:
Busy Tone:
Reorder Tone:
Off Hook Warning Tone:
Ring Back Tone:
Confirm Tone:
SIT1 Tone:
SIT2 Tone:
SIT3 Tone:
SIT4 Tone:
MWI Dial Tone:
Cfwd Dial Tone:
Holding Tone:
Conference Tone:
Secure Call Indication Tone:
VoIP PIN Tone:
PSTN PIN Tone:
Feature Invocation Tone:

Distinctive Ring Patterns
Ring1 Cadence: Ring2 Cadence:
Ring3 Cadence: Ring4 Cadence:
Ring5 Cadence: Ring6 Cadence:
Ring7 Cadence: Ring8 Cadence:

Distinctive Call Waiting Tone Patterns
CWT1 Cadence: CWT2 Cadence:
CWT3 Cadence: CWT4 Cadence:
CWT5 Cadence: CWT6 Cadence:
CWT7 Cadence: CWT8 Cadence:

Distinctive Ring/CWT Pattern Names
Ring1 Name: Ring2 Name:
Ring3 Name: Ring4 Name:
Ring5 Name: Ring6 Name:
Ring7 Name: Ring8 Name:

Ring and Call Waiting Tone Spec
Ring Waveform: SinusoidTrapezoid Ring Frequency:
Ring Voltage: CWT Frequency:

Control Timer Values (sec)
Hook Flash Timer Min: Hook Flash Timer Max:
Callee On Hook Delay: Reorder Delay:
Call Back Expires: Call Back Retry Intvl:
Call Back Delay: VMWI Refresh Intvl:
Interdigit Long Timer: Interdigit Short Timer:
CPC Delay: CPC Duration:

Vertical Service Activation Codes
Call Return Code: Call Redial Code:
Blind Transfer Code: Call Back Act Code:
Call Back Deact Code: Call Back Busy Act Code:
Cfwd All Act Code: Cfwd All Deact Code:
Cfwd Busy Act Code: Cfwd Busy Deact Code:
Cfwd No Ans Act Code: Cfwd No Ans Deact Code:
Cfwd Last Act Code: Cfwd Last Deact Code:
Block Last Act Code: Block Last Deact Code:
Accept Last Act Code: Accept Last Deact Code:
CW Act Code: CW Deact Code:
CW Per Call Act Code: CW Per Call Deact Code:
Block CID Act Code: Block CID Deact Code:
Block CID Per Call Act Code: Block CID Per Call Deact Code:
Block ANC Act Code: Block ANC Deact Code:
DND Act Code: DND Deact Code:
CID Act Code: CID Deact Code:
CWCID Act Code: CWCID Deact Code:
Dist Ring Act Code: Dist Ring Deact Code:
Speed Dial Act Code: Secure All Call Act Code:
Secure No Call Act Code: Secure One Call Act Code:
Secure One Call Deact Code: Conference Act Code:
Attn-Xfer Act Code: Modem Line Toggle Code:
FAX Line Toggle Code: Media Loopback Code:
Referral Services Codes:
Feature Dial Services Codes:

Vertical Service Announcement Codes
Service Annc Base Number:
Service Annc Extension Codes:

Outbound Call Codec Selection Codes
Prefer G711u Code: Force G711u Code:
Prefer G711a Code: Force G711a Code:
Prefer G723 Code: Force G723 Code:
Prefer G726r16 Code: Force G726r16 Code:
Prefer G726r24 Code: Force G726r24 Code:
Prefer G726r32 Code: Force G726r32 Code:
Prefer G726r40 Code: Force G726r40 Code:
Prefer G729a Code: Force G729a Code:

Miscellaneous
Set Local Date (mm/dd): Set Local Time (HH/mm):
Time Zone: GMT-12:00GMT-11:00GMT-10:00GMT-09:00GMT-08:00GMT-07:00GMT-06:00GMT-05:00GMT-04:00GMT-03:30GMT-03:00GMT-02:00GMT-01:00GMTGMT+01:00GMT+02:00GMT+03:00GMT+03:30GMT+04:00GMT+05:00GMT+05:30GMT+05:45GMT+06:00GMT+06:30GMT+07:00GMT+08:00GMT+09:00GMT+09:30GMT+10:00GMT+11:00GMT+12:00GMT+13:00 FXS Port Impedance: 600900600+2.16uF900+2.16uF270+750||150nF220+820||120nF220+820||115nF200+600||100nF
Daylight Saving Time Rule:
FXS Port Input Gain: FXS Port Output Gain:
DTMF Playback Level: DTMF Playback Length:
Detect ABCD: yesno Playback ABCD: yesno
Caller ID Method: Bellcore(N.Amer,China)DTMF(Finland,Sweden)DTMF(Denmark)ETSI DTMFETSI DTMF With PRETSI DTMF After RingETSI FSKETSI FSK With PR(UK)DTMF(Denmark) With PR Caller ID FSK Standard: bell 202v.23
Feature Invocation Method: DefaultSweden More Echo Suppression: yesno



Line Enable: yesno

Streaming Audio Server (SAS)
SAS Enable: yesno SAS DLG Refresh Intvl:
SAS Inbound RTP Sink:

NAT Settings
NAT Mapping Enable: yesno NAT Keep Alive Enable: yesno
NAT Keep Alive Msg: NAT Keep Alive Dest:

Network Settings
SIP ToS/DiffServ Value: SIP CoS Value: [0-7]
RTP ToS/DiffServ Value: RTP CoS Value: [0-7]
Network Jitter Level: lowmediumhighvery highextremely high Jitter Buffer Adjustment: up and downup onlydown onlydisable

SIP Settings
SIP Transport: UDPTCPTLS SIP Port:
SIP 100REL Enable: yesno EXT SIP Port:
Auth Resync-Reboot: yesno SIP Proxy-Require:
SIP Remote-Party-ID: yesno SIP GUID: yesno
SIP Debug Option: none1-line1-line excl. OPT1-line excl. NTFY1-line excl. REG1-line excl. OPT|NTFY|REGfullfull excl. OPTfull excl. NTFYfull excl. REGfull excl. OPT|NTFY|REG RTP Log Intvl:
Restrict Source IP: yesno Referor Bye Delay:
Refer Target Bye Delay: Referee Bye Delay:
Refer-To Target Contact: yesno Sticky 183: yesno
Auth INVITE: yesno

Call Feature Settings
Blind Attn-Xfer Enable: yesno MOH Server:
Xfer When Hangup Conf: yesno

Proxy and Registration
Proxy:
Outbound Proxy:
Use Outbound Proxy: yesno Use OB Proxy In Dialog: yesno
Register: yesno Make Call Without Reg: yesno
Register Expires: Ans Call Without Reg: yesno
Use DNS SRV: yesno DNS SRV Auto Prefix: yesno
Proxy Fallback Intvl: Proxy Redundancy Method: NormalBased on SRV Port
Voice Mail Server: Mailbox Subscribe Expires:

Subscriber Information
Display Name: User ID:
Password: Use Auth ID: yesno
Auth ID:
Mini Certificate:
SRTP Private Key:

Supplementary Service Subscription
Call Waiting Serv: yesno Block CID Serv: yesno
Block ANC Serv: yesno Dist Ring Serv: yesno
Cfwd All Serv: yesno Cfwd Busy Serv: yesno
Cfwd No Ans Serv: yesno Cfwd Sel Serv: yesno
Cfwd Last Serv: yesno Block Last Serv: yesno
Accept Last Serv: yesno DND Serv: yesno
CID Serv: yesno CWCID Serv: yesno
Call Return Serv: yesno Call Redial Serv: yesno
Call Back Serv: yesno Three Way Call Serv: yesno
Three Way Conf Serv: yesno Attn Transfer Serv: yesno
Unattn Transfer Serv: yesno MWI Serv: yesno
VMWI Serv: yesno Speed Dial Serv: yesno
Secure Call Serv: yesno Referral Serv: yesno
Feature Dial Serv: yesno Service Announcement Serv: yesno

Audio Configuration
Preferred Codec: G711uG711aG726-16G726-24G726-32G726-40G729aG723 Silence Supp Enable: yesno
Use Pref Codec Only: yesno Silence Threshold: highmediumlow
G729a Enable: yesno Echo Canc Enable: yesno
G723 Enable: yesno Echo Canc Adapt Enable: yesno
G726-16 Enable: yesno Echo Supp Enable: yesno
G726-24 Enable: yesno FAX CED Detect Enable: yesno
G726-32 Enable: yesno FAX CNG Detect Enable: yesno
G726-40 Enable: yesno FAX Passthru Codec: G711uG711a
DTMF Process INFO: yesno FAX Codec Symmetric: yesno
DTMF Process AVT: yesno FAX Passthru Method: NoneNSEReINVITE
DTMF Tx Method: InBandAVTINFOAutoInBand+INFOAVT+INFO DTMF Tx Mode: NormalStrict
FAX Process NSE: yesno Hook Flash Tx Method: NoneAVTINFO
FAX Disable ECAN: yesno Release Unused Codec: yesno
FAX Enable T38: yesno FAX T38 Redundancy: 0123
FAX Tone Detect Mode: caller or calleecaller onlycallee only Symmetric RTP: yesno

Gateway Accounts
Gateway 1: GW1 NAT Mapping Enable: yesno
GW1 Auth ID: GW1 Password:
Gateway 2: GW2 NAT Mapping Enable: yesno
GW2 Auth ID: GW2 Password:
Gateway 3: GW3 NAT Mapping Enable: yesno
GW3 Auth ID: GW3 Password:
Gateway 4: GW4 NAT Mapping Enable: yesno
GW4 Auth ID: GW4 Password:

VoIP Fallback To PSTN
Auto PSTN Fallback: yesno

Dial Plan
Dial Plan:
Enable IP Dialing: yesno Emergency Number:

FXS Port Polarity Configuration
Idle Polarity: ForwardReverse Caller Conn Polarity: ForwardReverse
Callee Conn Polarity: ForwardReverse



Line Enable: yesno

NAT Settings
NAT Mapping Enable: yesno NAT Keep Alive Enable: yesno
NAT Keep Alive Msg: NAT Keep Alive Dest:

Network Settings
SIP ToS/DiffServ Value: SIP CoS Value: [0-7]
RTP ToS/DiffServ Value: RTP CoS Value: [0-7]
Network Jitter Level: lowmediumhighvery highextremely high Jitter Buffer Adjustment: up and downup onlydown onlydisable

SIP Settings
SIP Transport: UDPTCPTLS SIP Port:
SIP 100REL Enable: yesno EXT SIP Port:
Auth Resync-Reboot: yesno SIP Proxy-Require:
SIP Remote-Party-ID: yesno SIP GUID: yesno
SIP Debug Option: none1-line1-line excl. OPT1-line excl. NTFY1-line excl. REG1-line excl. OPT|NTFY|REGfullfull excl. OPTfull excl. NTFYfull excl. REGfull excl. OPT|NTFY|REG RTP Log Intvl:
Restrict Source IP: yesno Referor Bye Delay:
Refer Target Bye Delay: Referee Bye Delay:
Refer-To Target Contact: yesno Sticky 183: yesno
Auth INVITE: yesno

Proxy and Registration
Proxy:
Outbound Proxy:
Use Outbound Proxy: yesno Use OB Proxy In Dialog: yesno
Register: yesno Make Call Without Reg: yesno
Register Expires: Ans Call Without Reg: yesno
Use DNS SRV: yesno DNS SRV Auto Prefix: yesno
Proxy Fallback Intvl: Proxy Redundancy Method: NormalBased on SRV Port

Subscriber Information
Display Name: User ID:
Password: Use Auth ID: yesno
Auth ID:
Mini Certificate:
SRTP Private Key:

Audio Configuration
Preferred Codec: G711uG711aG726-16G726-24G726-32G726-40G729aG723 Silence Supp Enable: yesno
Use Pref Codec Only: yesno Echo Canc Enable: yesno
G729a Enable: yesno Echo Canc Adapt Enable: yesno
G723 Enable: yesno Echo Supp Enable: yesno
G726-16 Enable: yesno FAX CED Detect Enable: yesno
G726-24 Enable: yesno FAX CNG Detect Enable: yesno
G726-32 Enable: yesno FAX Passthru Codec: G711uG711a
G726-40 Enable: yesno FAX Codec Symmetric: yesno
DTMF Process INFO: yesno FAX Passthru Method: NoneNSEReINVITE
DTMF Process AVT: yesno DTMF Tx Method: InBandAVTINFOAutoInBand+INFOAVT+INFO
DTMF Tx Mode: NormalStrict Release Unused Codec: yesno
FAX Process NSE: yesno Symmetric RTP: yesno
FAX Disable ECAN: yesno

Dial Plans
Dial Plan 1:
Dial Plan 2:
Dial Plan 3:
Dial Plan 4:
Dial Plan 5:
Dial Plan 6:
Dial Plan 7:
Dial Plan 8:

VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable: yesno VoIP Caller Auth Method: nonePINHTTP Digest
VoIP PIN Max Retry: One Stage Dialing: yesno
Line 1 VoIP Caller DP: none12345678 VoIP Caller Default DP: none12345678
Line 1 Fallback DP: none12345678
VoIP Caller ID Pattern:
VoIP Access List:
VoIP Caller 1 PIN: VoIP Caller 1 DP: none12345678
VoIP Caller 2 PIN: VoIP Caller 2 DP: none12345678
VoIP Caller 3 PIN: VoIP Caller 3 DP: none12345678
VoIP Caller 4 PIN: VoIP Caller 4 DP: none12345678
VoIP Caller 5 PIN: VoIP Caller 5 DP: none12345678
VoIP Caller 6 PIN: VoIP Caller 6 DP: none12345678
VoIP Caller 7 PIN: VoIP Caller 7 DP: none12345678
VoIP Caller 8 PIN: VoIP Caller 8 DP: none12345678

VoIP Users and Passwords (HTTP Authentication)
VoIP User 1 Auth ID: VoIP User 1 DP: none12345678
VoIP User 1 Password:
VoIP User 2 Auth ID: VoIP User 2 DP: none12345678
VoIP User 2 Password:
VoIP User 3 Auth ID: VoIP User 3 DP: none12345678
VoIP User 3 Password:
VoIP User 4 Auth ID: VoIP User 4 DP: none12345678
VoIP User 4 Password:
VoIP User 5 ID Auth ID: VoIP User 5 DP: none12345678
VoIP User 5 Password:
VoIP User 6 Auth ID: VoIP User 6 DP: none12345678
VoIP User 6 Password:
VoIP User 7 Auth ID: VoIP User 7 DP: none12345678
VoIP User 7 Password:
VoIP User 8 Auth ID: VoIP User 8 DP: none12345678
VoIP User 8 Password:

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yesno PSTN Caller Auth Method: nonePIN
PSTN Ring Thru Line 1: yesno PSTN PIN Max Retry:
PSTN CID For VoIP CID: yesno PSTN CID Number Prefix:
PSTN Caller Default DP: 12345678 Off Hook While Calling VoIP: yesno
Line 1 Signal Hook Flash To PSTN: DisabledDouble Hook Flash PSTN CID Name Prefix:
PSTN Caller ID Pattern:
PSTN Access List:
PSTN Caller 1 PIN: PSTN Caller 1 DP: 12345678
PSTN Caller 2 PIN: PSTN Caller 2 DP: 12345678
PSTN Caller 3 PIN: PSTN Caller 3 DP: 12345678
PSTN Caller 4 PIN: PSTN Caller 4 DP: 12345678
PSTN Caller 5 PIN: PSTN Caller 5 DP: 12345678
PSTN Caller 6 PIN: PSTN Caller 6 DP: 12345678
PSTN Caller 7 PIN: PSTN Caller 7 DP: 12345678
PSTN Caller 8 PIN: PSTN Caller 8 DP: 12345678

FXO Timer Values (sec)
VoIP Answer Delay: VoIP PIN Digit Timeout:
PSTN Answer Delay: PSTN PIN Digit Timeout:
PSTN-To-VoIP Call Max Dur: PSTN Ring Thru Delay:
VoIP-To-PSTN Call Max Dur: PSTN Ring Thru CWT Delay:
VoIP DLG Refresh Intvl: PSTN Ring Timeout:
PSTN Dialing Delay: PSTN Dial Digit Len:
PSTN Hook Flash Len:

PSTN Disconnect Detection
Detect CPC: yesno Detect Polarity Reversal: yesno
Detect PSTN Long Silence: yesno Detect VoIP Long Silence: yesno
PSTN Long Silence Duration: VoIP Long Silence Duration:
PSTN Silence Threshold: very highhighmediumlowvery low Min CPC Duration:
Detect Disconnect Tone: yesno
Disconnect Tone:

International Control
FXO Port Impedance: 600900270+750||150nF220+820||120nF370+620||310nF320+1050||230nF370+820||110nF275+780||115nF120+820||110nF350+1000||210nF0+900||30nF600+2.16uF900+1uF900+2.16uF600+1uFGlobal Ring Frequency Min:
SPA To PSTN Gain: Ring Frequency Max:
PSTN To SPA Gain: Ring Validation Time: 100 ms150 ms200 ms256 ms384 ms512 ms640 ms1024 ms
Tip/Ring Voltage Adjust: 3.1 V3.2 V3.35 V3.5 V Ring Indication Delay: 0256 ms512 ms768 ms1024 ms1280 ms1536 ms1792 ms
Operational Loop Current Min: 10 mA12 mA14 mA16 mA Ring Timeout: 0128 ms256 ms384 ms512 ms640 ms768 ms896 ms1024 ms1152 ms1280 ms1408 ms1536 ms1664 ms1792 ms1920 ms
On-Hook Speed: Less than 0.5 ms3 ms (ETSI)26 ms (Australia) Ring Threshold: 13.5-16.5 Vrms19.35-23.65 Vrms40.5-49.5 Vrms
Current Limiting Enable: yesno Ringer Impedance: High (Normal)Synthesized (Poland,S.Africa,Slovenia)
Line-In-Use Voltage:


Call Forward Settings
Cfwd All Dest: Cfwd Busy Dest:
Cfwd No Ans Dest: Cfwd No Ans Delay:

Selective Call Forward Settings
Cfwd Sel1 Caller: Cfwd Sel1 Dest:
Cfwd Sel2 Caller: Cfwd Sel2 Dest:
Cfwd Sel3 Caller: Cfwd Sel3 Dest:
Cfwd Sel4 Caller: Cfwd Sel4 Dest:
Cfwd Sel5 Caller: Cfwd Sel5 Dest:
Cfwd Sel6 Caller: Cfwd Sel6 Dest:
Cfwd Sel7 Caller: Cfwd Sel7 Dest:
Cfwd Sel8 Caller: Cfwd Sel8 Dest:
Cfwd Last Caller: Cfwd Last Dest:
Block Last Caller: Accept Last Caller:

Speed Dial Settings
Speed Dial 2: Speed Dial 3:
Speed Dial 4: Speed Dial 5:
Speed Dial 6: Speed Dial 7:
Speed Dial 8: Speed Dial 9:

Supplementary Service Settings
CW Setting: yesno Block CID Setting: yesno
Block ANC Setting: yesno DND Setting: yesno
CID Setting: yesno CWCID Setting: yesno
Dist Ring Setting: yesno Secure Call Setting: yesno
Message Waiting: yesno Accept Media Loopback Request: neverautomaticmanual
Media Loopback Mode: sourcemirror Media Loopback Type: mediapacket

Distinctive Ring Settings
Ring1 Caller: Ring2 Caller:
Ring3 Caller: Ring4 Caller:
Ring5 Caller: Ring6 Caller:
Ring7 Caller: Ring8 Caller:

Ring Settings
Default Ring: 12345678 Default CWT: 12345678
Hold Reminder Ring: 12345678none Call Back Ring: 12345678
Cfwd Ring Splash Len: Cblk Ring Splash Len:
VMWI Ring Splash Len: VMWI Ring Policy: New VM AvailableNew VM Becomes AvailableNew VM Arrives
Ring On No New VM: yesno


PSTN-To-VoIP Selective Call Forward Settings
Cfwd Sel1 Caller: Cfwd Sel1 Dest:
Cfwd Sel2 Caller: Cfwd Sel2 Dest:
Cfwd Sel3 Caller: Cfwd Sel3 Dest:
Cfwd Sel4 Caller: Cfwd Sel4 Dest:
Cfwd Sel5 Caller: Cfwd Sel5 Dest:
Cfwd Sel6 Caller: Cfwd Sel6 Dest:
Cfwd Sel7 Caller: Cfwd Sel7 Dest:
Cfwd Sel8 Caller: Cfwd Sel8 Dest:

PSTN-To-VoIP Speed Dial Settings
Speed Dial 2: Speed Dial 3:
Speed Dial 4: Speed Dial 5:
Speed Dial 6: Speed Dial 7:
Speed Dial 8: Speed Dial 9:

PSTN Ring Thru Line 1 Distinctive Ring Settings
Ring1 Caller: Ring2 Caller:
Ring3 Caller: Ring4 Caller:
Ring5 Caller: Ring6 Caller:
Ring7 Caller: Ring8 Caller:

PSTN Ring Thru Line 1 Ring Settings
Default Ring: 12345678Follow Line 1



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[invisage]

Sorry for the delay. ok I have worked it out. I know this sounds insane, through trail and error. i think it was the port setting are reversed on the 3102. will work it out exactly in the morning and post the config settings.

 

[invisage]

Here are copies of my 3102 configuration that now can both make and receive calls.
The 3CX 3102 guide http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE seems to be out off date. The defaults have obviously have changed.

 

[invisage]

ANThe working config

 

[techworx]

congratulations on working this out, i am just stuck on what you mean about a standard ata port on 3cx config , how do you config this please. sorry but im a newbie to this.

 

[invisage]

this configuration uses a stardard phone as a 3CX extension on the 3102. just disregard these settings if you dont need the extension.

 

[h2009]

I just tried to configure my 3102 after all this time, and im having issues with it. I've done it as its described by 3CX, and using the changes mentioned on here. Bridge mode seems to kill the unit.

Can i ask what firmware verisons is everybody running?

It seems that if call the PSTN line, all it does is ring to the caller, but the calls are not getting passed to the PBX, and if i try to make a call out on the PSTN line the call is failed. I'll post a log up ASAP.

 

[techworx]

thanks a lot mate ,working a treat. i was just about to get the hammer at it

 

[Henk]

I posted a link to a manual a while back, that should supply most of the details you need.

Try a search on my name, for sure you will find an answer. If not let me know.

H.