Hello List,<div><br></div><div>I have been trying to setup T38 gatewaying with the following setup</div><div><br></div><div>SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO</div><div><br></div><div>I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo Canceller: HWEC</div>
<div><br></div><div>I'm aware there's no support for T38 gateway but I have been trying to get the patches <a href="https://issues.asterisk.org/view.php?id=13405">https://issues.asterisk.org/view.php?id=13405</a> to work. It seems like some people have been able to get transparent T38 gateway to work on 1.8</div>
<div><br></div><div>I have downloaded the latest patch <meta http-equiv="content-type" content="text/html; charset=utf-8"><span class="Apple-style-span" style="font-family: Verdana, Arial, Helvetica, sans-serif; font-size: 13px; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px; "><a href="https://issues.asterisk.org/file_download.php?file_id=28947&type=bug" target="_blank">asterisk-1.8.4_fax.patch</a> applied it. Installed my spandsp spandsp-0.0.6pre18.tgz and it compiles succesfully.</span></div>
<div><font class="Apple-style-span" face="Verdana, Arial, Helvetica, sans-serif"><span class="Apple-style-span" style="-webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br></span></font></div>
<div><font class="Apple-style-span" face="Verdana, Arial, Helvetica, sans-serif"><span class="Apple-style-span" style="-webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;">I'm using zoiper to test the T38 faxing. Here is my dialplan<br clear="all">
</span></font><br></div><div><div> '_5062XXXXXXX' => 1. Set(FAXOPT(t38gateway)=yes) [pbx_config]</div><div> 2. Dial(DAHDI/g0/${EXTEN}) [pbx_config]</div><div>
3. Hangup() [pbx_config]</div><div><br></div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [50624309954@termination-test:1] Set("SIP/robert-00000007", "GROUP(customers)=ibasis") in new stack</div>
<div> -- Executing [50624309954@termination-test:2] Set("SIP/robert-00000007", "GROUP(termination)=ss7") in new stack</div><div> -- Executing [50624309954@termination-test:3] Set("SIP/robert-00000007", "FAXOPT(t38gateway)=yes") in new stack</div>
<div> -- Executing [50624309954@termination-test:4] Dial("SIP/robert-00000007", "DAHDI/g0/50624309954") in new stack</div><div> -- Called g0/50624309954</div><div> -- DAHDI/1-1 is proceeding passing it to SIP/robert-00000007</div>
<div>[Apr 8 00:26:35] NOTICE[31251]: chan_sip.c:8615 process_sdp: T.38 re-INVITE detected but no fax extension</div><div> -- Running Gateway activestate=4 (SIP/robert-00000007) and inactivestate=0 (DAHDI/1-1)</div><div>
[Apr 8 00:26:35] ERROR[10783]: res_fax.c:822 fax_session_reserve: Could not locate a FAX technology module with capabilities (T38_GATEWAY)</div><div>[Apr 8 00:26:35] ERROR[10783]: res_fax.c:2527 __ast_t38_gateway_handle_parameters: Unable to reserve FAX session.</div>
<div> -- Running Gateway activestate=0 (DAHDI/1-1) and inactivestate=4 (SIP/robert-00000007)</div><div>[Apr 8 00:26:35] ERROR[10783]: res_fax.c:822 fax_session_reserve: Could not locate a FAX technology module with capabilities (T38_GATEWAY)</div>
<div>[Apr 8 00:26:35] ERROR[10783]: res_fax.c:2527 __ast_t38_gateway_handle_parameters: Unable to reserve FAX session.</div><div><br></div><div>The call gets established but the fax on the other side, receive nothing and disconnect. The reciving fax complains about timeout.</div>
<div><br></div><div>I added the faxdetect app created by the patch. I asked the reported about what it actually does as we don't know anything further than is related to the T38 kickover</div><div><br></div><div> '_5062XXXXXXX' => 1. Set(FAXOPT(t38gateway)=yes) [pbx_config]</div>
<div> 2. FaxDetect(5) [pbx_config]</div><div> 3. Dial(DAHDI/g0/${EXTEN}) [pbx_config]</div><div> 4. Hangup() [pbx_config]</div>
<div><br></div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [50624309954@termination-test:1] Set("SIP/robert-00000005", "GROUP(customers)=ibasis") in new stack</div><div> -- Executing [50624309954@termination-test:2] Set("SIP/robert-00000005", "GROUP(termination)=ss7") in new stack</div>
<div> -- Executing [50624309954@termination-test:3] Set("SIP/robert-00000005", "FAXOPT(t38gateway)=yes") in new stack</div><div> -- Executing [50624309954@termination-test:4] FaxDetect("SIP/robert-00000005", "5") in new stack</div>
<div> -- Executing [50624309954@termination-test:5] Dial("SIP/robert-00000005", "DAHDI/g0/50624309954") in new stack</div><div> -- Called g0/50624309954</div><div> -- DAHDI/1-1 is proceeding passing it to SIP/robert-00000005</div>
<div> -- DAHDI/1-1 answered SIP/robert-00000005</div><div><br></div><div>But then the call drop after 5 seconds with an 488 not acceptable here. </div><div><br></div><div><meta http-equiv="content-type" content="text/html; charset=utf-8"><div>
o=ZoiperCommunicator_user 0 1 IN IP4 192.168.1.60</div><div>s=ZoiperCommunicator_session</div><div>c=IN IP4 192.168.1.60</div><div>t=0 0</div><div>m=image 8000 udptl t38</div><div>a=T38FaxVersion:0</div><div>a=T38MaxBitRate:14400</div>
<div>a=T38FaxFillBitRemoval:0</div><div>a=T38FaxTranscodingMMR:0</div><div>a=T38FaxTranscodingJBIG:0</div><div>a=T38FaxRateManagement:transferredTCF</div><div>a=T38FaxMaxBuffer:400</div><div>a=T38FaxMaxDatagram:400</div><div>
a=T38FaxUdpEC:t38UDPRedundancy</div></div><div><br></div><div>Asterisk returns</div><div><br></div><div><div>SIP/2.0 488 Not acceptable here</div><div>Via: SIP/2.0/UDP 192.168.1.60:53325;branch=z9hG4bK-d8754z-35a890bc09845fb2-1---d8754z-;received=201.192.28.178;rport=53325</div>
<div>From: "robert"<<a href="mailto:sip%3Arobert@190.106.66.210">sip:robert@190.106.66.210</a>;transport=UDP>;tag=0905b421</div><div>To: <<a href="mailto:sip%3A50624309954@190.106.66.210">sip:50624309954@190.106.66.210</a>;transport=UDP>;tag=as605221e3</div>
<div>Call-ID: NGU5YzQ5NDdiNTU3ZDg1N2VmYTM3MGQwNDhlMjlkNGE.</div><div>CSeq: 2 INVITE</div><div>Server: Asterisk PBX 1.8.3.2</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>
Supported: replaces, timer</div><div>Content-Length: 0</div></div><div><br></div><div>I can see this type of behaviour with T38 is quite common</div><div><br></div><div><a href="https://issues.asterisk.org/view.php?id=16327">https://issues.asterisk.org/view.php?id=16327</a></div>
<div><meta http-equiv="content-type" content="text/html; charset=utf-8"><a href="https://issues.asterisk.org/view.php?id=16793">https://issues.asterisk.org/view.php?id=16793</a></div><div><meta http-equiv="content-type" content="text/html; charset=utf-8"><a href="https://issues.asterisk.org/view.php?id=16793">https://issues.asterisk.org/view.php?id=16793</a></div>
<div><br></div><div>I was wondering if anyone has some experience with this patch, or T38 asterisk transparent gateway.</div></div><div><br></div><div>I think the problem could be more on the sip channel rejecting the call, can someone help me try to narrow down the issue. I have more captures and logs available </div>
<div><br></div><div>-- <br>Robert<br>
</div>