[soundar]

Hi Everyone
I am trying to make PSTN -> VOIP work for the last one week and i am unable to make it work.
I use a SPA 3000 : firware version 3.1.10d

I am able to dial my PSTN number, i get the PIN TONE, i input my pin and I get the VOIP dial tone. I can dial a outward number.... then i get reorder tone.

I see the the info page i can see that the PSTN line is connected, the peer number appears ( that is the number i dialed) it says calling. the RTP port number appears, all the parameters like packet sent , recvd, latency get activated and all show zero. no movement in the parameters. and a while later FAILED TO CONNECT.

According to me SPA recognises the VOIP number i dial, because it appears on the peer number, but it is not dialling as the packets sent and received is zero. And why SPA is telling VOIP CALL FAILED, without even dialing ?

I am using VOIP buster on Line 1 and the same on PSTN page too.

I live in Jakarta, and my PSTN provider does not provide caller ID service. Could this be the problem that my PSTN to VOIP is not working. I have set the PSTN CID for VOIP CID "NO"

The info page shows PSTN CALLER : , (just one comma or quotation mark is not clear )

I tries setting Anonymous in CALLER id patter or in the PSTN caller access list... still cannot get through the PSTN to VOIP.

can some body plese help me understand the complexity and make PSTN TO VOIP work for me

thanks in advance

regds

soundar

[rizsher]

No, CallerID is not a must.. it does make things simpler, but the PIN number setup you've got works just as well.

The reason why the calls don't complete could be for a number of reason, maybe you've got the same voip provider configured on both Line 1 and PSTN tab VoIP setup, and the provider doesnt allow double registration, maybe you haven't setup any provider on PSTN... do a search on the Linksys forum here, this has been discussed to death here...

If oyu still don't reach a resolution, post your config html pages here (also disucssed here how thats done) and someone can have a look and possibly help you out.

[soundar]

Thank you for your immediate reply and encouragement to a new user like me.

Before posting to the forum i did a lot of research not only voxilla forums but other forums too... tried implementing most of the suggestions but still no avail.

i am attaching the configs ...

I am using a WRT54GS router with DD-WRT v27 firmware.

What i am confused is the calling number appears on the info page... then the msg CALL FAILED

[miroesq]

Hey Soundar,

Since your using the same account on both Line 1 and the PSTN Line, make sure that in your PSTN tab, you have register set to no and make calls without registering set to yes. I think that should take care of it.

Good luck

[soundar]

I have already set Line1 to VOIPBUSTER and PSTN to GIZMO..still the same problem....
when i call the info page shows the number being called. I checked the serverlog... it shows it is being dialed. then after abt 30 seconds it shows CALL FAILED. I captured the syslog.... i cannot understand most of it. But there appears a line ICMP ERROR -1.... has it got something to do.. but if i call from Line1 the call is alright...

thanks in advance

soundar

[soundar]

I tried to capture the syslog entries and i am getting an ICMP ERROR -1 , please GURU explain the following entries and tell me what should i do to PSTN hop on to voip to work

Calling:18004672513@sip1.voipbuster.com:0
Oct 9 08:54:42 000E08CB84C5 [1:0]AUD ALLOC CALL (port=16456)
Oct 9 08:54:42 000E08CB84C5 [1:0]RTP Rx Up
Oct 9 08:54:42 000E08CB84C5 RSE_DEBUG: reference domain:sip1.voipbuster.com
Oct 9 08:54:42 000E08CB84C5 [1]SIP:ICMP Error -1 (50efebc8:5060, 3)
Oct 9 08:54:42 000E08CB84C5 RSE_DEBUG: getting alternate from domain:sip1.voipbuster.com
Oct 9 08:54:43 000E08CB84C5 [1]SIP:ICMP Error -1 (50efebc9:5060, 3)
Oct 9 08:54:43 000E08CB84C5 RSE_DEBUG: getting alternate from domain:sip1.voipbuster.com
Oct 9 08:54:43 000E08CB84C5 [1]SIP:ICMP Error -1 (c27800ca:5060, 3)
Oct 9 08:54:43 000E08CB84C5 RSE_DEBUG: getting alternate from domain:sip1.voipbuster.com
Oct 9 08:54:44 000E08CB84C5 [1]SIP:ICMP Error -1 (c27800cb:5060, 3)
Oct 9 08:54:44 000E08CB84C5 RSE_DEBUG: getting alternate from domain:sip1.voipbuster.com
Oct 9 08:54:44 000E08CB84C5 [1]SIP:ICMP Error -1 (c2dd3ece:5060, 3)
Oct 9 08:54:44 000E08CB84C5 RSE_DEBUG: getting alternate from domain:sip1.voipbuster.com
Oct 9 08:54:44 000E08CB84C5 [1:0]AUD Rel Call
Oct 9 08:54:44 000E08CB84C5 CC:Failed

[rizsher]

Config html file??

[soundar]

Below is the current config

Product Name: SPA3000
Serial Number: 88012DA66660
Software Version: 3.1.10(GWd)
Hardware Version: 3.0.0(9607)
Enable Web Server: yes
Category : Internet Connection Type
DHCP: yes
Static IP:
NetMask:
Gateway:
Category : Optional Network Configuration
HostName:
Domain: voipbuster.com
Primary DNS:
Secondary DNS:
DNS Server Order: Manual
DNS Query Mode: Parallel
Syslog Server: 192.168.x.x:514
Debug Server: 192.168.x.x:514
Debug Level: 3+H
Primary NTP Server:
Secondary NTP Server:
Page : SIP
Category : SIP Parameters
Max Forward: 70
Max Redirection: 5
Max Auth: 2
SIP User Agent Name: $VERSION
SIP Server Name: $VERSION
SIP Reg User Agent Name:
SIP Accept Language:
DTMF Relay MIME Type: application/dtmf-relay
Hook Flash MIME Type: application/hook-flash
Remove Last Reg: no
Use Compact Header: no
Escape Display Name: no
Category : SIP Timer Values (sec)
SIP T1: .5
SIP T2: 4
SIP T4: 5
SIP Timer B: 32
SIP Timer F: 32
SIP Timer H: 32
SIP Timer D: 32
SIP Timer J: 32
INVITE Expires: 240
ReINVITE Expires: 30
Reg Min Expires: 1
Reg Max Expires: 7200
Reg Retry Intvl: 30
Reg Retry Long Intvl: 1200
Category : Response Status Code Handling
SIT1 RSC:
SIT2 RSC:
SIT3 RSC:
SIT4 RSC:
Try Backup RSC:
Retry Reg RSC:
Category : RTP Parameters
RTP Port Min: 16384
RTP Port Max: 16482
RTP Packet Size: 0.030
Max RTP ICMP Err: 0
RTCP Tx Interval: 0
No UDP Checksum: no
Stats In BYE: no
Category : SDP Payload Types
NSE Dynamic Payload: 100
AVT Dynamic Payload: 101
INFOREQ Dynamic Payload:
G726r16 Dynamic Payload: 98
G726r24 Dynamic Payload: 97
G726r32 Dynamic Payload: 2
G726r40 Dynamic Payload: 96
G729b Dynamic Payload: 99
NSE Codec Name: NSE
AVT Codec Name: telephone-event
G711u Codec Name: PCMU
G711a Codec Name: PCMA
G726r16 Codec Name: G726-16
G726r24 Codec Name: G726-24
G726r32 Codec Name: G726-32
G726r40 Codec Name: G726-40
G729a Codec Name: G729a
G729b Codec Name: G729ab
G723 Codec Name: G723
Category : NAT Support Parameters
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Test Enable: no
STUN Server: stun.sipdiscount.com
EXT IP:
EXT RTP Port Min:
NAT Keep Alive Intvl: 20
Page : Regional
all Regional tabs is set to defaults I did not change
Category : Ring and Call Waiting Tone Spec
Ring Waveform: Sinusoid
Ring Frequency: 25
Ring Voltage: 70
CWT Frequency: 440@-10
Category : Control Timer Values (sec)
Hook Flash Timer Min: .1
Hook Flash Timer Max: .9
Callee On Hook Delay: 0
Reorder Delay: 5
Call Back Expires: 1800
Call Back Retry Intvl: 30
Call Back Delay: .5
VMWI Refresh Intvl: 0
Interdigit Long Timer: 10
Interdigit Short Timer: 3
CPC Delay: 2
CPC Duration: 0
Category : Vertical Service Activation Codes
all set to defaults
Time Zone: GMT-08:00
FXS Port Impedance: 600
Daylight Saving Time Rule: start=4/-1/7;end=10/1/7;save=1
FXS Port Input Gain: -3
FXS Port Output Gain: -3
DTMF Playback Level: -16
DTMF Playback Length: .1
Detect ABCD: yes
Playback ABCD: yes
Caller ID Method: Bellcore(N.Amer,China)
FXS Port Power Limit: 3
Caller ID FSK Standard: bell 202
Feature Invocation Method: Default
Page : Line 1
Category : Line 1
Line Enable: yes
Category : Streaming Audio Server (SAS)
SAS Enable: no
SAS DLG Refresh Intvl: 30
SAS Inbound RTP Sink:
Category : NAT Settings
NAT Mapping Enable: no
NAT Keep Alive Enable: no
NAT Keep Alive Msg: $NOTIFY
NAT Keep Alive Dest: $PROXY
Category : Network Settings
SIP TOS/DiffServ Value: 0x68
Network Jitter Level: high
RTP TOS/DiffServ Value: 0xb8
Jitter Buffer Adjustment: up and down
Category : SIP Settings
SIP Port: 5060
SIP 100REL Enable: no
EXT SIP Port:
Auth Resync-Reboot: yes
SIP Proxy-Require:
SIP Remote-Party-ID: yes
SIP GUID: no
SIP Debug Option: full
RTP Log Intvl: 0
Restrict Source IP: no
Referor Bye Delay: 4
Refer Target Bye Delay: 0
Referee Bye Delay: 0
Refer-To Target Contact: no
Sticky 183: no
Category : Call Feature Settings
Blind Attn-Xfer Enable: no
MOH Server:
Xfer When Hangup Conf: yes
Category : Proxy and Registration
Proxy: fwd.pulver.com
Use Outbound Proxy: no
Outbound Proxy:
Use OB Proxy In Dialog: yes
Register: yes
Make Call Without Reg: no
Register Expires: 3600
Ans Call Without Reg: no
Use DNS SRV: no
DNS SRV Auto Prefix: no
Proxy Fallback Intvl: 3600
Proxy Redundancy Method: Normal
Voice Mail Server:
Mailbox Subscribe Expires: 2147483647
Category : Subscriber Information
Display Name: soundaxxxxxx
User ID: 80xxxx
Password: *************
Use Auth ID: no
Auth ID:
Mini Certificate:
SRTP Private Key:
Category : Supplementary Service Subscription
**** DEFAULTS *******
Category : Audio Configuration
Preferred Codec: G711u
Silence Supp Enable: no
Use Pref Codec Only: no
Silence Threshold: medium
G729a Enable: yes
Echo Canc Enable: yes
G723 Enable: yes
Echo Canc Adapt Enable: yes
G726-16 Enable: yes
Echo Supp Enable: yes
G726-24 Enable: yes
FAX CED Detect Enable: yes
G726-32 Enable: yes
FAX CNG Detect Enable: yes
G726-40 Enable: yes
FAX Passthru Codec: G711u
DTMF Process INFO: yes
FAX Codec Symmetric: yes
DTMF Process AVT: yes
FAX Passthru Method: NSE
DTMF Tx Method: Auto
FAX Process NSE: yes
Hook Flash Tx Method: None
FAX Disable ECAN: no
Release Unused Codec: yes
Symmetric RTP: yes
Category : Gateway Accounts
Gateway 1:
GW1 NAT Mapping Enable: no
GW1 Auth ID:
GW1 Password:
Gateway 2:
GW2 NAT Mapping Enable: no
GW2 Auth ID:
GW2 Password:
Gateway 3:
GW3 NAT Mapping Enable: no
GW3 Auth ID:
GW3 Password:
Gateway 4:
GW4 NAT Mapping Enable: no
GW4 Auth ID:
GW4 Password:
Category : VoIP Fallback To PSTN
Auto PSTN Fallback: yes
Category : Dial Plan
Dial Plan: (*xx|xxxxxxxxxxxx.)
Enable IP Dialing: no
Emergency Number:
Category : FXS Port Polarity Configuration
Idle Polarity: Forward
Caller Conn Polarity: Forward
Callee Conn Polarity: Forward

PSTN TAB ON NEXT POST

________________________________________
Please Note :
On the pstn tab i have tried Register Yes and also No
I have tried with and without Outbound proxy, Nat keep alive, Nat enable, etc etc.

Even onthe SIP Tab too, i have tried all type of permutation and combination handle VIA received, rport etc.

The question of ICMP Error -1 many people have complained earlier in the forum, but to my limited search it seems there has been no clear reply from the members so far. Correct me if i am wrong.

Thanks for the encouragement

regds

soundar

[soundar]

CONFIG CONTINUED

Page : PSTN Line
Category : PSTN Line
Line Enable: yes
Category : NAT Settings
NAT Mapping Enable: yes
NAT Keep Alive Enable: yes
NAT Keep Alive Msg: $REGISTER
NAT Keep Alive Dest: $PROXY
Category : Network Settings
SIP TOS/DiffServ Value: 0x68
Network Jitter Level: high
RTP TOS/DiffServ Value: 0xb8
Jitter Buffer Adjustment: up and down
Category : SIP Settings
SIP Port: 5061
SIP 100REL Enable: no
EXT SIP Port:
Auth Resync-Reboot: yes
SIP Proxy-Require:
SIP Remote-Party-ID: yes
SIP GUID: no
SIP Debug Option: full
RTP Log Intvl: 0
Restrict Source IP: yes
Referor Bye Delay: 4
Refer Target Bye Delay: 0
Referee Bye Delay: 0
Refer-To Target Contact: no
Sticky 183: no
Category : Proxy and Registration
Proxy: sip1.voipbuster.com
Use Outbound Proxy: yes
Outbound Proxy: sip1.voipbuster.com
Use OB Proxy In Dialog: yes
Register: yes
Make Call Without Reg: yes
Register Expires: 3600
Ans Call Without Reg: yes
Use DNS SRV: no
DNS SRV Auto Prefix: no
Proxy Fallback Intvl: 3600
Proxy Redundancy Method: Normal
Category : Subscriber Information
Display Name: soundar
User ID: *********
Password: *************
Use Auth ID: no
Auth ID:
Mini Certificate:
SRTP Private Key:
Category : Audio Configuration
Preferred Codec: G711u
Silence Supp Enable: no
Use Pref Codec Only: no
Echo Canc Enable: yes
G729a Enable: yes
Echo Canc Adapt Enable: yes
G723 Enable: yes
Echo Supp Enable: yes
G726-16 Enable: yes
FAX CED Detect Enable: yes
G726-24 Enable: yes
FAX CNG Detect Enable: yes
G726-32 Enable: yes
FAX Passthru Codec: G711u
G726-40 Enable: yes
FAX Codec Symmetric: yes
DTMF Process INFO: yes
FAX Passthru Method: NSE
DTMF Process AVT: yes
DTMF Tx Method: Auto
Release Unused Codec: yes
FAX Process NSE: yes
Symmetric RTP: yes
FAX Disable ECAN: no
Warn Outgoing PSTN Call: no
Category : Dial Plans
Dial Plan 1: (xx.)
Dial Plan 2: (xx.)
Dial Plan 3: (xx.)
Dial Plan 4: (xx.)
Dial Plan 5: (xx.)
Dial Plan 6: (xx.)
Dial Plan 7: (xx.)
Dial Plan 8: (xx.)
Category : VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: none
VoIP PIN Max Retry: 3
One Stage Dialing: yes
Line 1 VoIP Caller DP: 1
VoIP Caller Default DP: 1
Line 1 Fallback DP: none
VoIP Caller ID Pattern:
VoIP Access List:
VoIP Caller 1 PIN:
VoIP Caller 1 DP: 1
VoIP Caller 2 PIN:
VoIP Caller 2 DP: 1
VoIP Caller 3 PIN:
VoIP Caller 3 DP: 1
VoIP Caller 4 PIN:
VoIP Caller 4 DP: 1
VoIP Caller 5 PIN:
VoIP Caller 5 DP: 1
VoIP Caller 6 PIN:
VoIP Caller 6 DP: 1
VoIP Caller 7 PIN:
VoIP Caller 7 DP: 1
VoIP Caller 8 PIN:
VoIP Caller 8 DP: 1
Category : VoIP Users and Passwords (HTTP Authentication)
VoIP User 1 Auth ID:
VoIP User 1 DP: 1
VoIP User 1 Password:
VoIP User 2 Auth ID:
VoIP User 2 DP: 1
VoIP User 2 Password:
VoIP User 3 Auth ID:
VoIP User 3 DP: 1
VoIP User 3 Password:
VoIP User 4 Auth ID:
VoIP User 4 DP: 1
VoIP User 4 Password:
VoIP User 5 ID Auth ID:
VoIP User 5 DP: 1
VoIP User 5 Password:
VoIP User 6 Auth ID:
VoIP User 6 DP: 1
VoIP User 6 Password:
VoIP User 7 Auth ID:
VoIP User 7 DP: 1
VoIP User 7 Password:
VoIP User 8 Auth ID:
VoIP User 8 DP: 1
VoIP User 8 Password:
Category : PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: yes
PSTN PIN Max Retry: 3
PSTN CID For VoIP CID: no
PSTN CID Number Prefix:
PSTN Caller Default DP: 8
Off Hook While Calling VoIP: no
e 1 Signal Hook Flash To PSTN: Disabled
PSTN CID Name Prefix:
PSTN Caller ID Pattern:
PSTN Access List:
PSTN Caller 1 PIN:
PSTN Caller 1 DP: 1
PSTN Caller 2 PIN:
PSTN Caller 2 DP: 1
PSTN Caller 3 PIN:
PSTN Caller 3 DP: 1
PSTN Caller 4 PIN:
PSTN Caller 4 DP: 1
PSTN Caller 5 PIN:
PSTN Caller 5 DP: 1
PSTN Caller 6 PIN: *************
PSTN Caller 6 DP: 6
PSTN Caller 7 PIN:
PSTN Caller 7 DP: 1
PSTN Caller 8 PIN:
PSTN Caller 8 DP: 1
Category : FXO Timer Values (sec)
VoIP Answer Delay: 0
VoIP PIN Digit Timeout: 10
PSTN Answer Delay: 2
PSTN PIN Digit Timeout: 10
PSTN-To-VoIP Call Max Dur: 0
PSTN Ring Thru Delay: 1
VoIP-To-PSTN Call Max Dur: 0
PSTN Ring Thru CWT Delay: 3
VoIP DLG Refresh Intvl: 0
PSTN Ring Timeout: 5
PSTN Dialing Delay: 1
PSTN Dial Digit Len: .1/.1
PSTN Hook Flash Len: .25
Category : PSTN Disconnect Detection
Detect CPC: yes
Detect Polarity Reversal: yes
Detect PSTN Long Silence: no
Detect VoIP Long Silence: no
PSTN Long Silence Duration: 30
VoIP Long Silence Duration: 30
PSTN Silence Threshold: medium
Min CPC Duration: 0.2
Detect Disconnect Tone: yes
Disconnect Tone: 480@-30,620@-30;4(.25/.25/1+2)
Category : International Control
FXO Port Impedance: Global
Ring Frequency Min: 10
SPA To PSTN Gain: 0
Ring Frequency Max: 100
PSTN To SPA Gain: 0
Ring Validation Time: 256 ms
Tip/Ring Voltage Adjust: 3.5 V
Ring Indication Delay: 512 ms
Operational Loop Current Min: 10 mA
Ring Timeout: 640 ms
On-Hook Speed: Less than 0.5 ms
Ring Threshold: 13.5-16.5 Vrms
Current Limiting Enable: no
Ringer Impedance: High (Normal)
Line-In-Use Voltage: 30

[Twix]

Hi All,

New here and I have a simular case as stated.

Setup.

ISP --> Modem (ADSL2) has two pstn phone lines (Is actually voip over ISP network but acts as pstn phone line)

From the same modem there is UTP line that goes to my LINUX box which acts as a FW / DHCP server.

DHCP server provides local IP's to clients in the network.

All of the above work great.

I also have an SPA3102. It works great as well but I can't seem to get both CID and the pstn gateway to work. let me explain.

In the modem I can choose 2 CID modes. One is FSK and the other is DTMF. My DECT phone works on DTMF. So, if I set the modem to DTMF I get caller ID going to my DECT and that all works fine. If I now use my mobile phone and based on CID from it I have my PSTN GATEWAY set up, the spa does not give me the authentication beeps and rings through to the to the DECT phone. I.O. I ring my house, the spa sees my mobile phone number but does not connect me to the VOIP gateway.

Here is the mind boggle. If I dont change anything in the SPA but set my modem to FSK, the SPA does recognize my mobile number, it connects me to the authentication and after entering the PIN to the VOIP gateway. However with these setting I don't have CID on my DECT because it does not work with FSK.

It seems that if my modem is set to FSK the SPA recognizes the phone number before it picks up and if the modem is set to DTMF the SPA recognizes CID after it picks up.

Any suggestions on how to have both work.


Cheers Twix

PS i'm in the Netherlands