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<p>Hi. the fax show version does not work since i am not using the
digium modem.</p>
<p>the iax2 show peers is the command for me and the output is:</p>
<p>PBX*CLI> iax2 show peers<br>
Name/Username Host Mask
Port Status Description<br>
iaxmodem/iaxmod 127.0.0.1 (S) 255.255.255.255
4570 OK (1 ms)<br>
1 iax2 peers [1 online, 0 offline, 0 unmonitored]<br>
PBX*CLI><br>
</p>
<p><br>
</p>
<p>the problem is that in logs i am getting errors and i do not know
how to fix it.</p>
<p>root@PBX: /var/log/iaxmodem $ more ttyIAX0<br>
[2016-11-16 09:08:12.483144] Registration failed.<br>
[2016-11-16 09:13:05.118692] Terminating on signal 15...<br>
[2016-11-16 09:21:49.181872] Registration failed.<br>
[2016-11-16 09:22:30.731893] Terminating on signal 15...<br>
[2016-11-16 09:22:30.759221] Registration failed.<br>
[2016-11-16 09:25:11.014642] Registration failed.<br>
root@PBX: /var/log/iaxmodem $<br>
</p>
<p><br>
</p>
<p><br>
</p>
<p>Any ideas?</p>
<p><br>
</p>
<p><br>
</p>
<br>
<div class="moz-cite-prefix">On 15/11/2016 5:40 μμ, Larry Moore
wrote:<br>
</div>
<blockquote
cite="mid:4efd363f-c8e4-46d8-5cdd-cebed07cfe14@moorecomputing.local"
type="cite">
<meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
I suspect I followed a guide much like the one you have used
including information found on voip-info - sorry, I can't seem to
find any bookmarks of relevant information.<br>
<br>
I spent an enormous amount of time getting it working and working
very well, the real issue was getting T.38 working - I applied a
patch to Asterisk version 1.8 to get the T.38 gateway
functionality.<br>
<br>
I would have started off my testing by confirming communications
between two IAX modems, I presume you are using HylaFAX too.<br>
<br>
Once the communications between the two IAX modems was working I
progressed with testing sending and receiving faxes using G711A
through my VoIP service and a modem attached to a PSTN service,
suffice to say T.38 functionality was the key to getting reliable
faxes working through VoIP at least when traversing the Internet,
fortunately my VoIP provider facilitates T.38.<br>
<br>
Using an SPA8800 on my network I tested sending and receiving
faxes with a modem attached to the SPA8800, it worked in G711A and
T.38.<br>
<br>
I progressed to Asterisk 11 where the T.38 gateway functionality
is better along with other improvements.<br>
<br>
What is the output on your system for:<br>
<br>
fax show version<br>
<br>
<br>
Cheers,<br>
<br>
Larry.<br>
<br>
<div class="moz-cite-prefix">On 15/11/2016 8:09 PM, tux john
wrote:<br>
</div>
<blockquote
cite="mid:trinity-a1663328-acd3-45ba-ac15-7241cc84eb11-1479211745735@msvc-mesg-gmxus002"
type="cite">
<meta name="viewport" content="width=device-width">
<meta http-equiv="Content-Type" content="text/html;
charset=UTF-8">
<div class="mail_android_message" style="line-height: 1;
padding: 0.5em">Hi. Since I am messing a lot with it without
seeing the end of, may I ask if there is any solid guide for
that please?<br>
</div>
<div class="mail_android_quote" style="line-height: 1; padding:
0.3em">On 13/11/2016, 07:42 Larry Moore <a
moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="mailto:lmoore@starwon.com.au"><lmoore@starwon.com.au></a>
wrote:
<blockquote class="gmail_quote" style="margin: 0.8ex 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;"> Some additional information which may
help you with your installation.<br>
<br>
I have 4 IAX Modems named iaxmodem0 - iaxmodem3. I use
iaxmodem3 for outbound fax transmissions.<br>
<br>
I created a queue for the other 3 modems, here is my entry
in queues.conf:<br>
<blockquote> [hylafax-iax]<br>
strategy=linear<br>
ringinuse=yes<br>
autopause=no<br>
retry=4<br>
timeout=5<br>
timeoutpriority=conf<br>
reportholdtime=no<br>
joinempty=strict<br>
leavewhenempty=strict<br>
musicclass=silence<br>
<br>
member => IAX2/iaxmodem2<br>
member => IAX2/iaxmodem1<br>
member => IAX2/iaxmodem0<br>
<br>
</blockquote>
In case you are wondering about the 'musicclass' I have
used, here is the section from musiconhold.conf, the actual
location of the files may be elsewhere on your system:<br>
<blockquote> [silence]<br>
mode=files<br>
directory=/usr/local/share/asterisk/silence<br>
; ls /usr/local/share/asterisk/silence<br>
; 10.gsm<br>
;<br>
; The file 10.gsm came from
/usr/local/share/asterisk/sounds/en/silence<br>
<br>
</blockquote>
I changed 'callbackextension' in my sip.conf for the trunk
so that it would go directly to the 'fax' extension in the
dialplan i.e. 'callbackextension=fax'.<br>
<br>
I've included the console output when an incoming fax is
received:<br>
<blockquote> == Using SIP RTP TOS bits 184<br>
-- Executing [fax@from-itsp:1]
NoOp("SIP/itsp-00000044", "Fax Detected 2016-11-13
12:33:40 +0800") in new stack<br>
-- Executing [fax@from-itsp:2]
GotoIf("SIP/itsp-00000044", "0?3:8") in new stack<br>
-- Goto (from-itsp,fax,8)<br>
-- Executing [fax@from-itsp:8]
NoOp("SIP/itsp-00000044", "Finish if_from-itsp_237") in
new stack<br>
-- Executing [fax@from-itsp:9]
GotoIf("SIP/itsp-00000044", "0?10:13") in new stack<br>
-- Goto (from-itsp,fax,13)<br>
-- Executing [fax@from-itsp:13]
NoOp("SIP/itsp-00000044", "Finish if_from-itsp_238") in
new stack<br>
-- Executing [fax@from-itsp:14]
Set("SIP/itsp-00000044", "FAXOPT(gateway)=yes") in new
stack<br>
-- Executing [fax@from-itsp:15]
Queue("SIP/itsp-00000044", "hylafax-iax,dRt,,,15") in new
stack<br>
-- Started music on hold, class 'silence', on
SIP/itsp-00000044<br>
-- Call accepted by 127.0.0.1 (format alaw)<br>
-- Format for call is (alaw)<br>
-- IAX2/iaxmodem2-3086 is ringing<br>
-- Stopped music on hold on SIP/itsp-00000044<br>
-- IAX2/iaxmodem2-3086 answered SIP/itsp-00000044<br>
> 0x89bac000 -- Probation passed - setting RTP
source address to <ITSP IP Address>:18998<br>
== Using UDPTL TOS bits 184<br>
-- Executing [h@from-itsp:1]
GotoIf("SIP/itsp-00000044", "0?2:3") in new stack<br>
-- Goto (from-itsp,h,3)<br>
-- Executing [h@from-itsp:3] NoOp("SIP/itsp-00000044",
"Finish if_from-itsp_239") in new stack<br>
-- Executing [h@from-itsp:4] NoOp("SIP/itsp-00000044",
"Call/Fax Ended 2016-11-13 12:36:41 +0800") in new stack<br>
-- Hungup 'IAX2/iaxmodem2-3086'<br>
== Spawn extension (from-itsp, fax, 15) exited non-zero
on 'SIP/itsp-00000044'<br>
<br>
</blockquote>
I'm sure you've already checked and confirmed you have
'alaw' and 'ulaw' codecs permitted in your IAX Modems,
iax.conf and sip.conf configurations<br>
<br>
To test your configuration you could set it up your
environment so that you send an outgoing fax to yourself
i.e. your dial your number at the VoIP provider, this
assumes when you dial your VoIP number a connection is made
back to you, you can then troubleshoot the communication.<br>
<br>
This is how I performed the majority of my tests.<br>
<br>
Not sure why you haven't explored the option of terminating
a fax call in Asterisk, you will need some scripts to
convert the received image to a PDF which is then e-mailed.
An offer was made to you to provide scripts, if you set this
up when your iaxmodem's aren't working a fallback will be
for Asterisk to accept the call as it falls through, one
thing you should know, if you use the T.38 Gateway in your
dialplan you will need to disabled it prior to Asterisk
terminating the call. I use extensions.ael so here is an
example, I've included the macro I use to receive a fax in
Asterisk:<br>
<blockquote> context from-itsp {<br>
<br>
s => {<br>
NoOp(Call Received ${STRFTIME(,,%F %T
%z)});<br>
Set(CHANNEL(language)=en_AU);<br>
Set(DIALTIMEOUT=30);<br>
Progress();<br>
NoOp(Call Received from ${CALLERID(name)},
Tel: ${CALLERID(num)});<br>
.<br>
. other conditions checked and extensions dialled<br>
.<br>
};<br>
<br>
fax => {<br>
NoOp(Fax Detected ${STRFTIME(,,%F %T
%z)});<br>
Set(FAXOPT(gateway)=yes);<br>
Queue(hylafax-iax,dRt,,,15);<br>
<br>
Set(FAXOPT(gateway)=no);<br>
&fax-receive(<TSID>,<Header>,FaxMaster,lmoore);<br>
Hangup();<br>
};<br>
<br>
h => {<br>
if ( "X${FAXRXFILE}" != "X" )<br>
{<br>
&email_rxfax();<br>
}<br>
NoOp(Call/Fax Ended ${STRFTIME(,,%F %T
%z)});<br>
};<br>
};<br>
<br>
macro fax-receive( fax-number, header-info, sender,
recipient ) {<br>
/*<br>
${ARG1} is Receiving Station Fax Number<br>
${ARG2} is Fax Header Information<br>
${ARG3} is Fax Sender E-mail Address<br>
${ARG4} is Fax Recipient E-mail Address<br>
*/<br>
NoOp(**** FAX RECEIVE ****);<br>
Set(FAXOPT(localstationid)=${LOCAL(fax-number)});<br>
Set(FAXOPT(headerinfo)=${LOCAL(header-info)});<br>
Set(FROMADDR=${LOCAL(sender)});<br>
Set(TOADDR=${LOCAL(recipient)});<br>
NoOp(**** SETTING FAXOPT ****);<br>
NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)});<br>
NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)});<br>
NoOp(FAXOPT(localstationid) :
${FAXOPT(localstationid)});<br>
Set(RXSTART=${EPOCH});<br>
Set(FAXRXPATH=/var/spool/asterisk/fax/received);<br>
Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});<br>
NoOp(**** RECEIVING FAX : ${FAXRXFILE} ****);<br>
ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f);<br>
NoOp(**** Subroutine Return ****);<br>
return;<br>
};<br>
</blockquote>
Cheers,<br>
<br>
Larry.<br>
<br>
<br>
<div class="moz-cite-prefix"> On 13/11/2016 8:07 AM, Larry
Moore wrote:<br>
</div>
<blockquote> Is your network/firewall configuration
permitting the ports for UDPTL, runn the command: udptl
show config<br>
<br>
<blockquote> UDPTL Global options<br>
--------------------<br>
udptlstart: 4000<br>
udptlend: 4999<br>
udptlfecentries: 3<br>
udptlfecspan: 3<br>
use_even_ports: No<br>
udptlchecksums: Yes<br>
<br>
</blockquote>
In your sip configuration for your 'mytrunk' peer have you
set applicable options e.g.:<br>
<blockquote> t38pt_udptl=yes,redundancy,maxdatagram=400<br>
</blockquote>
In your extensions.conf you could and probably should set
the following option prior to dialing the IAX channel,
this is to enable the T.38 gateway feature of Asterisk 11:<br>
<blockquote> Set(FAXOPT(gateway)=yes)<br>
</blockquote>
I have it working in my installation however I have
incoming voice calls too hence I use 'faxdetect' to direct
the call to the 'fax' extension.<br>
<br>
Cheers,<br>
<br>
Larry.<br>
<br>
<div class="moz-cite-prefix"> On 12/11/2016 5:24 AM, tux
john wrote:<br>
</div>
<blockquote>
<div style="font-family: Verdana;font-size: 12.0px;">
<div> hi. i am using asterisk 11.24.1 in my raspberry.
i do have a sip trunk with a provider with g711a. I
am trying to setup a fax server by following the
guide in<a moz-do-not-send="true"
class="moz-txt-link-freetext"
href="http://the-asterisk-book.com/1.6/faxserver.html"
target="_blank"> </a><a moz-do-not-send="true"
class="moz-txt-link-freetext"
href="http://the-asterisk-book.com/1.6/faxserver.html">http://the-asterisk-book.com/1.6/faxserver.html</a>.
</div>
<div> </div>
<div> i do live in Greece and the number is
00302112152130 </div>
<div> the problem is that i am getting the following
error and i am stuck: </div>
<div> </div>
<div>
<div> == Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [00302112152130@fax-in:1]
Dial("SIP/mytrunk-00000001", "IAX2/iaxmodem") in
new stack<br>
-- Called IAX2/iaxmodem<br>
-- Hungup 'IAX2/iaxmodem-3818'<br>
== Everyone is busy/congested at this time
(1:0/0/1)<br>
-- Auto fallthrough, channel
'SIP/mytrunk-00000001' status is 'CHANUNAVAIL'<br>
RasPBX*CLI> </div>
<div> </div>
<div> </div>
<div> the extensions.conf has </div>
<div>
<div> [fax-in]<br>
exten => 00302112152130,1,Dial(IAX2/iaxmodem)
</div>
<div> </div>
</div>
<div> </div>
<div> any ideas, please? </div>
</div>
</div>
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