[asterisk-users] T.38 not working - help needed with log interpretation

[Recursive]

Matt, thank you very much for your help!

> I'm just going to comment on the 'directmedia'/'canreinvite' points here.
> 
> 1) There is no 'reinvite' setting in chan_sip. If you patched
> Asterisk, than your mileage may vary.

I didn't patch. Just using "vanilla" Asterisk from your website ...

> 2) 'directmedia' is the same thing as 'canreinvite'. They are the same
> setting. 'directmedia' replaced the nomenclature 'canreinvite' because
> it actually describes what the setting does: it determines whether or
> not Asterisk will attempt to re-INVITE media directly between RTP
> capable peers.

This information is very valuable to me as it drastically cuts the number possible parameter combinations.

> 3) While documentation sometimes is lacking for some parts of
> Asterisk, this setting is actually pretty well documented in
> sip.conf.sample:

You are right. After compiling, I have made and installed the configuration sample files and have read those of them which I thought I needed. Thus, I already have read the section regarding directmedia, but I never would have come to the idea that this could be the same as canreinvite. Most examples around the net still use both without further explanation; even my ITSP sent me a sample configuration which also used both. Once again, thanks for clarifying!

> Note that none of this matters until you are in a bridge. If you are
> in a bridge, I would expect Asterisk to re-INVITE the media back to
> Asterisk when one of the sides offers T.38 (and, in fact, we have
> automated tests that check for this sort of thing). You shouldn't have
> to set directmedia to 'no', but - in the interest of making your
> system easier to debug and to remove variables - you may want to set
> it to 'no' for the involved peers.

I think I am in a bridge. As far as I can recall, I have seen respective messages in the Asterisk console after having started Asterisk with -vvvc, and as far as I have understood, there is no support for direct T38. I'll test again ...

> Just use 'directmedia'. They are the same setting (snippet from
> chan_sip's configuration parsing):
> 
>     } else if (!strcasecmp(v->name, "directmedia") ||
> !strcasecmp(v->name, "canreinvite")) {
>         ast_set_flag(&mask[0], SIP_REINVITE);
>         ...
 
Thanks again ... very instructive.

> Note that these settings and their behaviour is the same from 1.8
> through 13. While I'm glad to see anyone using the latest and greatest
> - yay Asterisk 13! - this isn't a reason to go to Asterisk 13.

For me, the reason was that I thought that I needed the gateway capability for faxing via T.38 (seems that I was wrong here), and that I didn't see any T.38 packet in the logs when using 1.8.x (regardless of which configuration parameters I was using).

Matt, I nearly don't dare to ask, but could you eventually take a quick look into the logs I have provided? Do you see any reason why asterisk hangs up, claiming a critical packet timeout, although all packets seemingly have been answered timely and appropriately?

Regards,

Recursive