On 03 May 2023, Gamgee said the following...

Hello all,

I am pleased to announce that Palantir BBS now has dialup/modem access! The phone number is: 850-332-1203
It's a voip line, using an ATA to connect the modem to, so it may not
be as stable/reliable as the old copper lines that we might remember;
but if you have a modem and a few minutes, I'd appreciate you giving it
a try! Will be monitoring and tweaking things if possible to improve
the service as we go along. Thanks!
--- SBBSecho 3.20-Linux
* Origin: Palantir * palantirbbs.ddns.net * Pensacola, FL * (21:2/138)

Gamgee, how did you configure your system to use your modem over voip?

I've been searching for a solution for days.

I've got a basic asterisk freepbx system with two pjsip exentions, and my cisco spa122 ata, but my modems can't complete the handshake. RING RING then beep but they never connect.

can you send me some knowledge nuggets?

... Why is the man who invests all your money called a broker?

--- Mystic BBS v1.12 A48 (Linux/64)
* Origin: The Bottomless Abyss BBS * bbs.bottomlessabyss.net (21:1/172)

bbsing wrote to Gamgee <=-

On 03 May 2023, Gamgee said the following...

I am pleased to announce that Palantir BBS now has dialup/modem access! The phone number is: 850-332-1203
It's a voip line, using an ATA to connect the modem to, so it may not
be as stable/reliable as the old copper lines that we might remember;
but if you have a modem and a few minutes, I'd appreciate you giving it
a try! Will be monitoring and tweaking things if possible to improve
the service as we go along. Thanks!

Gamgee, how did you configure your system to use your modem over voip?

I've been searching for a solution for days.

I've got a basic asterisk freepbx system with two pjsip exentions, and
my cisco spa122 ata, but my modems can't complete the handshake. RING
RING then beep but they never connect.

can you send me some knowledge nuggets?

Well, I've since stopped using it, since there were VERY few callers and
it just got to be a hassle. I see that you're using Mystic, and I'm
using Synchronet, so I don't know what differences that may introduce... There's a utility included with SBBS called 'sexpots' which is what I
was using. It "listens" to a serial port (or a USB port if you use a serial/usb adapter) and translates to TCP/IP for communication with the
BBS. I believe that it will also work with Mystic, as it's an external executable program that you could grab. The parameters are set up in 'sexpots.ini' - I will paste the contents of mine below. There is also
much more info on options/parameters here on the SBBS Wiki:

https://wiki.synchro.net/util:sexpots

Here is my .ini file, hope it helps and good luck!

LogLevel = INFO ; set display/log output level
Debug = TRUE ; enable debug logging (overrides LogLevel) PauseOnExit = FALSE ; wait for key-press on exit (non-service)
CLS = FALSE ; send a form feed (clear screen) before copyright banner
Prompt =
PromptTimeout = 30 ; seconds to wait for a remote character after sending prompt (0=infinite)

[COM]
Device = /dev/ttyUSB0 ; COM port device name (or port number)
BaudRate = 115200 ; If non-zero, use this DTE rate (e.g. 115200) Hangup = TRUE ; Hang-up phone after call
IgnoreDCD = FALSE ; Set to TRUE to ignore state of DCD
DCDTimeout = 10 ; Seconds to wait for DCD to drop
DTRDelay = 100 ; Milliseconds to delay before hangup
NullModem = FALSE ; Set to TRUE to not send AT commands to modem

[Modem]
Init = AT&F1 ; Modem initialization string
AutoAnswer = ATS0=1 ; Put modem into "auto-answer" mode
CleanUp = ATS0=0 ; When exiting, turn off auto-answer
EnableCallerID = AT#CID=1 ; Enable Caller-ID support (or try AT#CID=1) Timeout = 5 ; Seconds to wait for a response from modem
ReInit = 120 ; Minutes of inactivity while waiting for caller before re-initialization
Answer = ATA ; Answer command
Ring = RING ; Ring indication (from modem)
ManualAnswer = FALSE ; Set to TRUE to disable auto-answer and use ring detection/manual answer instead

[TCP]
Host = 192.168.254.70 ; Hostname or IP address of TCP server
Port = 23 ; TCP port number of TCP server
NoDelay = TRUE ; Set to TRUE to disable the Nagle Algorithm
Telnet = TRUE ; Set to FALSE to disable Telnet mode

[Telnet]
Debug = TRUE ; Set to TRUE to log Telnet commands sent/recv AdvertiseLocation = FALSE ; Set to TRUE to send "WILL SEND LOCATION"
TermType = SEXPOTS ; You shouldn't normally change this value TermSpeed = 115200,115200 ; Default terminal speed reported (tx, rx bps)

[Ident]
Enabled = FALSE ; Set to TRUE to enable Ident (RFC1413) server
Port = 113 ; TCP Port Ident server will listen on
Interface = 0 ; IP address of network interface (0=Any)
Response = CALLERID:SEXPOTS ; Resp-type and Add-info portions of response


... AAcckk!! II''mm iinn hhaallff dduupplleexx
=== MultiMail/Linux v0.52
--- SBBSecho 3.20-Linux
* Origin: Palantir * palantirbbs.ddns.net * Pensacola, FL * (21:2/138)

Gamgee, how did you configure your system to use your modem over voip

I've been searching for a solution for days.

I've got a basic asterisk freepbx system with two pjsip exentions, an my cisco spa122 ata, but my modems can't complete the handshake. RING RING then beep but they never connect.

can you send me some knowledge nuggets?

Well, I've since stopped using it, since there were VERY few callers and it just got to be a hassle. I see that you're using Mystic, and I'm using Synchronet, so I don't know what differences that may introduce... There's a utility included with SBBS called 'sexpots' which is what I
was using. It "listens" to a serial port (or a USB port if you use a serial/usb adapter) and translates to TCP/IP for communication with the BBS. I believe that it will also work with Mystic, as it's an external executable program that you could grab. The parameters are set up in 'sexpots.ini' - I will paste the contents of mine below. There is also much more info on options/parameters here on the SBBS Wiki:

https://wiki.synchro.net/util:sexpots

Here is my .ini file, hope it helps and good luck!

LogLevel = INFO ; set display/log output level
Debug = TRUE ; enable debug logging (overrides LogLevel) PauseOnExit = FALSE ; wait for key-press on exit (non-service)
CLS = FALSE ; send a form feed (clear screen) before copyright banner
Prompt =
PromptTimeout = 30 ; seconds to wait for a remote character
after sending prompt (0=infinite)

[COM]
Device = /dev/ttyUSB0 ; COM port device name (or port number) BaudRate = 115200 ; If non-zero, use this DTE rate (e.g.
115200) Hangup = TRUE ; Hang-up phone after call
IgnoreDCD = FALSE ; Set to TRUE to ignore state of DCD DCDTimeout = 10 ; Seconds to wait for DCD to drop
DTRDelay = 100 ; Milliseconds to delay before hangup NullModem = FALSE ; Set to TRUE to not send AT commands to
modem
[Modem]
Init = AT&F1 ; Modem initialization string
AutoAnswer = ATS0=1 ; Put modem into "auto-answer" mode
CleanUp = ATS0=0 ; When exiting, turn off auto-answer EnableCallerID = AT#CID=1 ; Enable Caller-ID support (or try AT#CID=1) Timeout = 5 ; Seconds to wait for a response from modem ReInit = 120 ; Minutes of inactivity while waiting for caller before re-initialization
Answer = ATA ; Answer command
Ring = RING ; Ring indication (from modem)
ManualAnswer = FALSE ; Set to TRUE to disable auto-answer and use ring detection/manual answer instead

[TCP]
Host = 192.168.254.70 ; Hostname or IP address of TCP server
Port = 23 ; TCP port number of TCP server
NoDelay = TRUE ; Set to TRUE to disable the Nagle Algorithm Telnet = TRUE ; Set to FALSE to disable Telnet mode

[Telnet]
Debug = TRUE ; Set to TRUE to log Telnet commands
sent/recv AdvertiseLocation = FALSE ; Set to TRUE to send "WILL SEND LOCATION" TermType = SEXPOTS ; You shouldn't normally change
this value TermSpeed = 115200,115200 ; Default terminal speed reported (tx, rx bps)
[Ident]
Enabled = FALSE ; Set to TRUE to enable Ident (RFC1413)
server Port = 113 ; TCP Port Ident server will listen on Interface = 0 ; IP address of network interface (0=Any) Response = CALLERID:SEXPOTS ; Resp-type and Add-info portions of response

Thank you Gamgee. That is some great information.

Where you using an ATA device to hook up your modem?

My problem is that the modems don't seem to connect, even they are set to ATS0=1. That handshake doesn't seem to be working. The modem picks up the line, then they screetch a bit but no completion of the handshake.

So I think I'm having some issues with the ATA or Asterisk, maybe both.

Asterisk is using ulaw, alaw for its codec. but for whatever reason modems just can't complete handshake.

If I can ever get the modems to connect during testing, then I can move to configuring a BBS to answer using them.

... Electricity is really just organized lightning.

--- Mystic BBS v1.12 A48 (Linux/64)
* Origin: The Bottomless Abyss BBS * bbs.bottomlessabyss.net (21:1/172)

bbsing wrote to Gamgee <=-

Gamgee, how did you configure your system to use your modem over voip

I've been searching for a solution for days.

I've got a basic asterisk freepbx system with two pjsip exentions, an my cisco spa122 ata, but my modems can't complete the handshake. RING RING then beep but they never connect.

can you send me some knowledge nuggets?

Well, I've since stopped using it, since there were VERY few callers and it just got to be a hassle. I see that you're using Mystic, and I'm
using Synchronet, so I don't know what differences that may introduce... There's a utility included with SBBS called 'sexpots' which is what I
was using. It "listens" to a serial port (or a USB port if you use a serial/usb adapter) and translates to TCP/IP for communication with the BBS. I believe that it will also work with Mystic, as it's an external executable program that you could grab. The parameters are set up in 'sexpots.ini' - I will paste the contents of mine below. There is also much more info on options/parameters here on the SBBS Wiki:

https://wiki.synchro.net/util:sexpots

Here is my .ini file, hope it helps and good luck!

<SNIP>

Thank you Gamgee. That is some great information.

Where you using an ATA device to hook up your modem?

Yes, a Grandstream HT-802. I did change a few default settings, that I
can't actually even remember now, but it wasn't too hard, and the info
was find-able online.

My problem is that the modems don't seem to connect, even they are set
to ATS0=1. That handshake doesn't seem to be working. The modem picks
up the line, then they screetch a bit but no completion of the
handshake.

So I think I'm having some issues with the ATA or Asterisk, maybe both.

The ATA can be quite finicky and sensitive to mis-configuration. I
don't have much knowledge on any of that any more.

Sorry I can't be more help... Hopefully you can figure it out. Seems
like it might be a modem initialization string issue.

Good luck!



... AAcckk!! II''mm iinn hhaallff dduupplleexx
=== MultiMail/Linux v0.52
--- SBBSecho 3.20-Linux
* Origin: Palantir * palantirbbs.ddns.net * Pensacola, FL * (21:2/138)

On 02 Sep 2024, Gamgee said the following...

bbsing wrote to Gamgee <=-

Gamgee, how did you configure your system to use your modem over

I've been searching for a solution for days.

I've got a basic asterisk freepbx system with two pjsip exention my cisco spa122 ata, but my modems can't complete the handshake. RING then beep but they never connect.
can you send me some knowledge nuggets?

Here is my .ini file, hope it helps and good luck!

<SNIP>

Thank you Gamgee. That is some great information.

Where you using an ATA device to hook up your modem?

Yes, a Grandstream HT-802. I did change a few default settings, that I can't actually even remember now, but it wasn't too hard, and the info was find-able online.

My problem is that the modems don't seem to connect, even they are se to ATS0=1. That handshake doesn't seem to be working. The modem picks up the line, then they screetch a bit but no completion of the handshake.

So I think I'm having some issues with the ATA or Asterisk, maybe bot

The ATA can be quite finicky and sensitive to mis-configuration. I
don't have much knowledge on any of that any more.

Sorry I can't be more help... Hopefully you can figure it out. Seems like it might be a modem initialization string issue.

Good luck!

Thanks Gamgee,

I sent out a message in the echos to anyone I searched who had information. Once I figure out who game the the link to the info I'll grab it.

But here is the info that got things going for me. ------------------------------------------------------------------

URL: https://gekk.info/articles/ata-config.html#Troubleshooting ===============================================================================
VoIP Setup

Once you have web access to the SPA as above, you can configure the ports. We will use the bogus number 9095551010, but if you want to use another one, just replace it (in yellow) below.

1. Log into the web interface

2. Go to the Voice section, then Line 1
1. Set Make Call Without Reg and Ans Call Without Reg to yes
2. Set User ID to 100

3. Scroll down and find Dialplan, and replace its contents with the following: o (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|<9095551010:101>S0<:@127.0.0.1:5061>|)

4. Under the Audio Configuration section, set everything that says Fax to no

5. Click Submit and wait about two minutes, then click on the Voice tab again if it doesn't redirect

3. Go to Line 2
1. Set Make Call Without Reg and Ans Call Without Reg to yes
2. Set User ID to 101
3. Scroll down and find Dialplan, and replace its contents with the following (it's different, so don't just reuse the first one!):

o (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|<9095551010:100>S0<:@127.0.0.1:5060>|)

4. Under the Audio Configuration section, set everything that says Fax to no

5. Click Submit and wait about two minutes, then click on the Voice tab again if it doesn't redirect

4. Configuration is complete!






===============================================================================
Troubleshooting

This should just work, but here are a couple things you can do if it doesn't: 1. Test basic dialing functionality:

1. Get a plain, basic telephone and plug it into one port

2. Try to dial 9095551010. Regardless of what's plugged into the other port, you should hear ringing. If you hear busy signal or dead air, you missed a config step.

3. If you have a second phone, plug it into the other port. Test dialing 9095551010 from either one; it should ring the other set and you should be able to pick up and talk.

4. If all of the above works then there's nothing wrong with the ATA dialing

2. Apply data optimization settings:

1. The instructions given earlier include the necessary step of disabling fax detection, but if that isn't enough, you can do this too.

2. In Line 1 and Line 2, apply the settings below. They will tell the ATA not to try to "help" and should cause it to just pass through audio unmodified.

1. After applying the settings to Line 1 and hitting Submit, make sure you wait for the page to reload before moving on to Line 2.

3. In the Network Settings section:
o Network Jitter Level: Extremely high
o Jitter Buffer Adjustment: No 4. In the Audio Configuration section:
o Preferred Codec: g711u
o Second and Third Preferred Codec: Unspecified
o G729a Enable: No
o Silence Supp Enable: No
o Echo Canc Enable: No o Everything that says Fax: No
o Modem Line: Yes

3. If you're not getting any dialtone, check that the SPA has an active Ethernet link on the blue port. If it doesn't have a connection and a valid IP, it'll shut off the voice module.

4. You will not get a 56k connection speed no matter what you do - the V.90 and V.92 specifications explicitly state that the modems you have ("analog modems") are only capable of originating a 33.6 connection. You need special ISP equipment to originate a 56k connection.


------------------------------------------------------------------

... If a pig loses its voice, is it disgruntled?

--- Mystic BBS v1.12 A48 (Linux/64)
* Origin: The Bottomless Abyss BBS * bbs.bottomlessabyss.net (21:1/172)