<div>Hello</div>
<div>Asterisk implement only passtrough T.38, so you cant terminate calls with asterisk using T.38.</div>
<div>You need T.38 gateways.</div>
<div> </div>
<div>Regards<br><br> </div>
<div><span class="gmail_quote">On 11/13/06, <b class="gmail_sendername">Ricardo Carvalho</b> <<a href="mailto:rjcarvalho@reit.up.pt">rjcarvalho@reit.up.pt</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Dear all,<br><br>I'm trying to enable Asterisk to work with FAX using T38. I've tried<br>Asterisk
1.2.4 with the available patch found at URL<br><a href="http://bugs.digium.com/view.php?id=5090">http://bugs.digium.com/view.php?id=5090</a> and also with the new 1.4 Beta3<br>that is announced to support it too.<br><br>With both Asterisk versions, I've sent with success FAXes between two
<br>FAX machines each one attached to an ATA interface, both registered in<br>Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as<br>far as I know, might be assigned with the "Content-Length" shown in the
<br>message header of every SIP message negotiating T38 parameters. I've<br>observed that after leaving Asterisk, the Content-Length of every<br>message carrying T38 parameters gets shorter than truly is, and<br>contrarily to my ATAs that seem to don't care about this, my Telco
<br>analyses the packet length written in this messages and truncates them,<br>aborting the call.<br><br>Does anyone experienced this too? Any ideas besides editing the<br>chan_sip.c code to fix this problem?<br><br>Thanks,
<br>Ricardo.<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:
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