Thanks. That's what I did, and it forced G.711, but the problem is that the fax failed anyway. Although a lot of RTP packets went through, the end result was an empty tif file :-(
Kevin, the F option is an excellent idea. Great. On Mon, Jun 27, 2011 at 10:14 PM, <[email protected]> wrote: > You could force g711 inbound by using > > Set(SIP_CODEC=ulaw) > -----Original Message----- > From: "Kevin P. Fleming" <[email protected]> > Sender: [email protected] > Date: Mon, 27 Jun 2011 14:08:00 > To: <[email protected]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] ReceiveFax to G.711 > > On 06/27/2011 08:06 AM, Michael wrote: > > > Controlling it through the sip.conf peers is sufficient for us for this > > case (because this particular provider doesn't support T.38 at all), but > > I think it would be a good idea to add the option to enable/disable T.38 > > from the dialplan. If I recall correctly, that's how callweaver worked > > at the time. > > In Asterisk trunk (soon to become Asterisk 1.10), there is an 'F' option > to ReceiveFAX and SendFAX that forces audio mode FAX even if the channel > is T.38 capable. That would do what you want. I posted a patch some time > ago for Asterisk 1.8 to add the same ability, but it probably doesn't > apply any more... the it's only a few lines though, it should be fairly > easy to replicate in the Asterisk 1.8 version of res_fax.c > > > Also, we just checked it, and since for that provider, we have other > > codecs in higher priorities (like GSM, for example) than G.711, G.711 > > was not chosen as the only codec, so the fax transmission failed. We can > > not prioritize G.711 over the other codecs in the sip.conf, for the > > obvious reasons, so for this, we need to do it in the dialplan. How can > > we do it? > > How are you going to determine that you need to force G.711 to be used? > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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