[resolved] Grandstream HT-502 and T.38

I have an HT-502 connecting my fax machine through Asterisk to Gafachi. T.38 with that setup, using Asterisk’s passthrough feature, works like a charm.

Today, I upgraded to Asterisk in order to do T.38 termination locally as well, but my luck there hasn’t been so great. G.711 from the ATA to * works, but I can’t even get the fax machine to connect with T.38, much less send pages. I get the following messages when I try:

[Jan 5 21:47:41] WARNING[18550]: app_fax.c:173 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely.
[Jan 5 21:47:41] WARNING[18550]: app_fax.c:618 transmit: Transmission error

I’m doing the recommended Answer then Wait for 3 seconds in order for the fax tone to signal * to use T.38, and I’ve put temporary debug messages in app_fax.c long enough to determine that T.38 negotiation is working. What I can’t figure out is why fax termination isn’t working.

Any ideas?

I’ve tried editing sip.conf according to the troubleshooting section of voip-info.org’s Asterisk T.38 page, but that didn’t work. I would try tweaking things in udptl.conf, but I’m not sure which direction to go there that would make the ATA work with Asterisk’s fax termination without breaking passthrough to Gafachi.

Apparently, I still have things to learn :smile:

It’s now working seamlessly in both cases, and the keys to that were apparently enabling reinvites for the ATA line connected to the fax machine, and setting the ATA’s fax tone detection mode to “callee” (I had it set to “both” before).


How does your set up look like ? Are you able to send or receive fax to/from PSTN in the following setup ?

Fax machine —ATA <------SIP------>Asterisk (T1/E1)<–PRI–>PSTN


Fax <-> ATA <-- SIP --> Asterisk <-- SIP --> VoIP/T.38 provider

I’ve never done anything as far as tying Asterisk more directly to the PSTN… mine is a pure VoIP/FoIP setup.

One thing I just noticed… Asterisk doesn’t seem to negotiate T.38 with the way I have things configured, so faxes sent to it are over G.711. That’s not that big a deal in my case, since the ATA and the PBX are both plugged into the same switch. At least it seems that T.38 is being used with Gafachi’s termination, though, according to what I’m seeing in network traffic after the re-invite.


Are you able to share the relevant extensions.conf and sip.conf for the following set up ?

Fax <-> ATA <-- SIP --> Asterisk <-- SIP --> VoIP/T.38 provider

In my setup of

Fax <-> ATA <-- SIP --> Asterisk <-- PRI—>PSTN

voice is working but I am unable to send and receive fax to and from PSTN.

In your setup are you first receiving the fax using app_fax and then sending the fax using app_fax to the final destination ?