Try open souce solution "ICTFAX" for T.38 faxing developed by us available at <a href="http://www.sourceforge.net/projects/ictfax">http://www.sourceforge.net/projects/ictfax</a><div><br></div><div><br clear="all">
Nasir Iqbal<br><br>ICT Innovations<br><a href="http://www.ictinnovations.com/" target="_blank">http://www.ictinnovations.com/</a><br><br>
<br><br><div class="gmail_quote">On Sat, Sep 4, 2010 at 3:03 AM, Joel Maslak <span dir="ltr"><<a href="mailto:jmaslak@antelope.net">jmaslak@antelope.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
g711 across a network without perfect jitter/delay characteristics will not work.<br><br>You cannot do g711 faxing across the internet - at all.<br><br>It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes).<div>
<div></div><div class="h5"><br>
<br><div class="gmail_quote">On Fri, Sep 3, 2010 at 12:32 PM, dave george <span dir="ltr"><<a href="mailto:dgeorge@teletoneinc.com" target="_blank">dgeorge@teletoneinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
Thanks Kevin,<br>
<br>
I tried passing it over VOIP using g711U codecs with no success. I will try<br>
using the patches that you mentioned and post the results.<br>
<div><br>
Thanks,<br>
Dave<br>
<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a><br>
</div><div>[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Kevin P.<br>
Fleming<br>
Sent: Friday, September 03, 2010 2:17 PM<br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Faxes<br>
<br>
</div><div><div></div><div>On 09/03/2010 10:50 AM, dave george wrote:<br>
> The asterisk box is connected to the PSTN using TE410 cards. Asterisk<br>
talk<br>
> SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the<br>
> PSTN.<br>
><br>
> The carrier sending the calls wants me to be able to pass faxes to<br>
physical<br>
> fax machines on the PSTN. So far they are failing.<br>
><br>
> We just want ot be able to pass faxes using g711u or t.38 pass through.<br>
<br>
As I told you on the asterisk-ss7 list, you can't 'pass through' T.38,<br>
because the PSTN does not speak T.38. If one side of the call is SIP,<br>
and the other side is TDM, then you have only two choices: pass the call<br>
through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX<br>
over T.38).<br>
<br>
At this time, the only option without patching Asterisk is to pass the<br>
call through in audio mode, but there are many, many problems with doing<br>
FAX over VoIP (Steve Underwood's page on the <a href="http://soft-switch.org" target="_blank">soft-switch.org</a> site<br>
explains them very well).<br>
<br>
There are patches in the issue tracker at <a href="http://issues.asterisk.org" target="_blank">issues.asterisk.org</a> to add<br>
T.38 gateway functionality to various releases of Asterisk, and they<br>
work well for quite a few people. If you added that, you'd be able to<br>
act as a T.38 gateway, which would dramatically increase your chances of<br>
success.<br>
<br>
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href="mailto:kfleming@digium.com" target="_blank">kfleming@digium.com</a><br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
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