g711 across a network without perfect jitter/delay characteristics will not work.
You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george <dgeo...@teletoneinc.com>wrote: > Thanks Kevin, > > I tried passing it over VOIP using g711U codecs with no success. I will > try > using the patches that you mentioned and post the results. > > Thanks, > Dave > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. > Fleming > Sent: Friday, September 03, 2010 2:17 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Faxes > > On 09/03/2010 10:50 AM, dave george wrote: > > The asterisk box is connected to the PSTN using TE410 cards. Asterisk > talk > > SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the > > PSTN. > > > > The carrier sending the calls wants me to be able to pass faxes to > physical > > fax machines on the PSTN. So far they are failing. > > > > We just want ot be able to pass faxes using g711u or t.38 pass through. > > As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, > because the PSTN does not speak T.38. If one side of the call is SIP, > and the other side is TDM, then you have only two choices: pass the call > through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX > over T.38). > > At this time, the only option without patching Asterisk is to pass the > call through in audio mode, but there are many, many problems with doing > FAX over VoIP (Steve Underwood's page on the soft-switch.org site > explains them very well). > > There are patches in the issue tracker at issues.asterisk.org to add > T.38 gateway functionality to various releases of Asterisk, and they > work well for quite a few people. If you added that, you'd be able to > act as a T.38 gateway, which would dramatically increase your chances of > success. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kflem...@digium.com > 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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