Hi,<br><br>I'm experiencing the same problem with t38modem and hylafax.<br>My problem is that on the re-Invite phase it syncs lower than 2400 bpps and the connection hangs on the second page.<br><br>Could you please post here the patch for asterisk 1.6.2.4 or even indicate which is the trunk of asterisk where this patch take effect?<br>
<br>thanks a lot,<br><br>Miguel Amez.<br><br><div class="gmail_quote">2010/5/3 Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">On 05/03/2010 11:59 AM, Ilmars Knipshis wrote:<br>
<br>
> Problem in short is as following:<br>
> after reINVITE from Cisco to negotiate T.38:<br>
><br>
> <--- SIP read from UDP:<a href="http://193.110.9.17:5060" target="_blank">193.110.9.17:5060</a> ---><br>
> INVITE <a href="mailto:sip%3A37166101111@159.148.78.220">sip:37166101111@159.148.78.220</a> SIP/2.0<br>
> Via: SIP/2.0/UDP <a href="http://193.110.9.17:5060" target="_blank">193.110.9.17:5060</a><br>
> From: <<a href="mailto:sip%3A3250890229@193.110.9.17">sip:3250890229@193.110.9.17</a>>;tag=74ff1200077fff10ff000018ff29ff16<br>
> To: "37166101111" <<a href="mailto:sip%3A37166101111@159.148.78.220">sip:37166101111@159.148.78.220</a>>;tag=as32fabaec<br>
> Call-ID: <a href="mailto:46ba3dad03495f6f35426980334703d3@159.148.78.220">46ba3dad03495f6f35426980334703d3@159.148.78.220</a><br>
> CSeq: 103 INVITE<br>
> Contact: <<a href="mailto:sip%3A3250890229@193.110.9.17">sip:3250890229@193.110.9.17</a>;user=phone><br>
> Max-Forwards: 10<br>
> User-Agent: MERA MSIP v.1.0.2<br>
> Content-Type: application/sdp<br>
> Content-Length: 183<br>
><br>
> v=0<br>
> o=- 1272610573 1272610573 IN IP4 193.110.9.17<br>
> s=-<br>
> c=IN IP4 193.110.9.17<br>
> t=0 0<br>
> m=image 25296 udptl t38<br>
> a=T38FaxRateManagement:transferredTCF<br>
> a=T38FaxUdpEC:t38UDPRedundancy<br>
<br>
</div>The problem here is being caused by the re-INVITE occurring prior to<br>
SendFAX() being started; this really should not be happening, as the<br>
other endpoint should not re-INVITE until it knows that a FAX endpoint<br>
is calling, but some of them do this anyway.<br>
<br>
There is a fix for this problem in SVN Asterisk trunk already, and it<br>
will be merged into the 1.6.2 branch in the next couple of weeks.<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">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>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>