----- "Kevin P. Fleming" <kpflem...@digium.com> escreveu: > Vinícius Fontes wrote: > > > I've put it on pastebin because is was a lot of text. Here's the > link: http://pastebin.com/m7467cea1. That's all the information on the > CLI with verbose=3 and "sip set debug peer voxip". > > OK, with the complete capture we can see that the problem is actually > quite different. In this call, Asterisk sent a re-INVITE to T.38 mode > from audio mode, the provider accepted it, and then Asterisk > acknowledged it. Immediately afterwards, Asterisk sent a re-INVITE > *back* to audio mode, which the provider accepted (and included T.38 > capabilities in their response). Because of this, the FAX reception > process failed since the T.38 session was destroyed. > > The most likely cause of this problem is a bug in chan_sip, but it > has > been fixed for quite some time now, and the fix is included in > 1.6.1.13. > This would also fit with your statement about not having this issue > with > Fax For Asterisk, as it does not generate any audio frames while > negotiating T.38 as the receiver of a FAX. > > I would suggest opening an issue in the issue tracker at > issues.asterisk.org and uploading your console trace there; there is > clearly a bug here that needs to be found and fixed. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org
Reported as issue 16576. Thanks for the support! Another related issue, and this one happens in FFA as well. I noticed I can only get really reliable fax reception if I edit chan_sip.c to force the bitrate down to 4800. Otherwise most of the times I get a few lines OK and the rest all garbage. Already talked to the telco tech support, they say there's no packet loss on their side (UDPTL isn't transmitted via Internet, my Asterisk box is connected to them using a dedicated VPN circuit), and I confirmed that with Wireshark. According to them, signalling is okay too. One thing I noticed is that Asterisk 1.6.1.13 completely ignores the maxdatagram setting on sip.conf. No matter what I set there, I keep getting the default value of 612 as offered by the telco. Asterisk not even once tries to negotiate that. Maybe (and that's a longshot) the datagram size is too long, or the buffers too low and Asterisk can't keep up with the reception of UDPTL packets? Is there any way to rule that out? As usual, I'm willing to provide any info to help solve this. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
