Thanks from the bottom of my heart sir!<br><br>Asterisk 1.8.7.1 is my build, I couldn't chose res_fax_spandsp from menuselect due to dependencies, so I installed libspandsp-dev:<br><br>:~# apt-cache policy libspandsp*<br>
libspandsp2:<br> Installed: 0.0.6~pre12-1<br> Candidate: 0.0.6~pre12-1<br> Version table:<br> *** 0.0.6~pre12-1 0<br><br>libspandsp-dev:<br> Installed: 0.0.6~pre12-1<br> Candidate: 0.0.6~pre12-1<br> Version table:<br>
*** 0.0.6~pre12-1 0<br><br>So, my dialplan snippet looks good, but I need to correct the lacking gateway capability?<br><br><br clear="all"> <b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"><font><font><span style="font-family:verdana,sans-serif"><span style="font-family:trebuchet ms,sans-serif"><span style="font-family:verdana,sans-serif">José Pablo Méndez</span></span></span></font><br>
</font></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><b style="color:rgb(153,153,153)"><span style="color:rgb(153,153,153)"></span></b><br>
<br><br><div class="gmail_quote">On Thu, Jan 5, 2012 at 10:04 AM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br>
<br>
Trying to set up res_fax_spandsp. Based on<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway" target="_blank">https://wiki.asterisk.org/<u></u>wiki/display/AST/T.38+Fax+<u></u>Gateway</a> I wrote this<br>
in my extensions.conf:<br>
<br>
exten => 306,1,NoOp(Fax transmission)<br>
same => n,Set(FAXOPT(gateway)=yes)<br>
same => n,Dial(DAHDI/3) ----->FXS port to fax machine<br>
same => n,Hangup()<br>
<br>
Call flow Im trying to pull out is as follows:<br>
<br>
Zoiper --> Asterisk with TDM410 --> FXS --> Analog fax machine<br>
<br>
I am totally lost about the use of this new gateway module in the<br>
dialplan. I think it loads ok:<br>
<br>
CLI> fax show capabilities<br>
<br>
Registered FAX Technology Modules:<br>
<br>
Type : Spandsp<br>
Description : Spandsp FAX Driver<br>
Capabilities : SEND RECEIVE T.38 G.711<br>
</blockquote>
<br></div>
Your res_fax_spandsp module was built without gateway support; what version of Asterisk are you using?<div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
1 registered modules<br>
<br>
Also I have the FFA manual, which I couldn't understand. I think FAXOPT<br>
is common to both, but still not sure how to put them together. Where<br>
can I find documentation about configuring the call flow described?<br>
</blockquote>
<br></div>
Fax for Asterisk is totally unrelated to T.38 gateway support.<div><div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Or some insight will also be appreciated.<br>
<br>
Here is my sip peer config:<br>
<br>
[105](headquarters) ;zoiper phone<br>
type=friend<br>
secret=<br>
mailbox=105@default<br>
t38pt_udptl = yes<br>
<br>
Dahdi:<br>
;FXS Modules<br>
group = 2<br>
signalling = fxo_ks<br>
context = interno<br>
channel = 3-4<br>
faxdetect = both<br>
<br>
Finally, a verbose output:<br>
<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [606@intern:1] NoOp("SIP/105-00000002", "Fax<br>
Transmission") in new stack<br>
-- Executing [606@intern:2] Set("SIP/105-00000002",<br>
"FAXOPT(gateway)=yes") in new stack<br>
[Jan 5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write:<br>
channel 'SIP/605-00000002' set FAXOPT(gateway) to 'yes' is unhandled!<br>
-- Executing [606@intern:3] Dial("SIP/605-00000002", "DAHDI/3") in<br>
new stack<br>
-- Called DAHDI/3<br>
-- DAHDI/3-1 is ringing<br>
-- DAHDI/3-1 is ringing<br>
-- DAHDI/3-1 is ringing<br>
-- DAHDI/3-1 answered SIP/605-00000002<br>
-- Hanging up on 'DAHDI/3-1'<br>
-- Hungup 'DAHDI/3-1'<br>
== Spawn extension (intern, 606, 3) exited non-zero on 'SIP/105-00000002'<br>
</blockquote>
<br></div></div>
Since 'fax show capabilities' did not show that you have gateway support, you won't be able to enable it. That problem must be corrected first.<span><font color="#888888"><br>
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href="mailto:kfleming@digium.com" target="_blank">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a> | Skype: kpfleming<br>
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</font></span></blockquote></div><br>