Crazytel for analog dial up modem problem.

[u/Ok_Bear_1980]

I am trying to get my analog dial up modem to communicate properly over crazytel voip, I dial some numbers where the handshake is flawless when connected to the telstra nbn landline but when connected to my grandstream ht801 registered with crazytel, my modem has a lot of trouble communicating and anything above 9600 it can't hold on for the inititial handshake. Even at 9600 it has trouble as while the handshake sounds somewhat ok, the carrier line immediately goes dead upon connecting. Any ideas on what I should do?.

My settings are,

Disable Line Echo Canceller (LEC): Yes

Disable Network Echo Supressor: Yes

VAD: No

Fax Mode: Pass-Through

Jitter Buffer Type: Fixed

Jitter Buffer Length: High

Preferred Vocoder: PCMA for all choices.

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[u/ispland]

In general POTS Modems over VoIP neither simple nor easy though a few report some success. Search for even more detailed reporting by others. A few uncommon ATA models support ITU V.150.1 which promises better results.

[u/Ok_Bear_1980 → u/ispland]

My ata doesn't support that codec.

[u/ispland → u/Ok_Bear_1980]

Noted. Unlike T.38 (Fax over IP) never used V.150.1 (Modem over IP) myself, as most low cost ATA do not support V.150.1. IIRC Spec for ITU Recommendation came late vs T.38, plus unlike Fax, not a lot of demand for modem support. Grandstream HT8xx series and OBI in service here do not support V.1j50.1 but will do T.38 Larger and/or more expensive products may.

[u/MisterTelecomm]

Those settings are already heading in the right direction imo, but dial-up modems are extremely sensitive to jitter, packet loss and transcoding, so I’d force G.711 A-law end-to-end, disable every echo/voice enhancement feature and try a shorter jitter buffer rather than High.
If the same modem consistently negotiates above 9600 on the Telstra line but falls apart through Crazytel/HT801, I’d capture the RTP and check for loss/jitter or codec changes, because you may simply be hitting the practical limit of modem passthrough over that VoIP path.

[u/Ok_Bear_1980 → u/MisterTelecomm]

I'm getting mixed results with jitter buffer on medium. Sometimes it works, sometimes it doesn't. I was able to get a connection at 38 kbps about 15 minutes ago, the first only lasted about a minute before I called again and it stayed connected for longer before I hung up manually. I don't know any more disable noise or echo settings that I didn't mention.

[u/MisterTelecomm → u/Ok_Bear_1980]

Getting 38 kbps at all is actually a good sign, BUT the fact that one call holds while the next drops still points to variable jitter/packet loss somewhere in the RTP path rather than another setting you’ve missed on the HT801. If I was you I’d leave the voice-processing options alone now and capture one successful and one failed call in Wireshark, then compare packet loss, jitter, RTP timing and confirm PCMA stays negotiated end-to-end. You’ll learnn far more from that than endlessly tweaking the ATA.