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text="#000000"><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 <lmoore@starwon.com.au> 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 class="moz-txt-link-freetext"
href="http://the-asterisk-book.com/1.6/faxserver.html"
target="_blank"></a><a
class="moz-txt-link-freetext"
href="http://the-asterisk-book.com/1.6/faxserver.html"
target="_blank">
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>
<br />
<fieldset
class="mimeAttachmentHeader"></fieldset><br />
</blockquote>
<br />
<br />
<fieldset
class="mimeAttachmentHeader"></fieldset><br />
</blockquote>
<br />
-- _____________________________________________________________________ --
Bandwidth and Colocation Provided by<a
href="http://www.api-digital.com" target="_blank">
http://www.api-digital.com</a> -- Check out the new Asterisk community
forum at:<a href="https://community.asterisk.org/"
target="_blank"> https://community.asterisk.org/</a> New to
Asterisk? Start here:<a
href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started"
target="_blank">
https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a>
asterisk-users mailing list To UNSUBSCRIBE or update options visit:<a
href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank">
http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</blockquote></div></body></html>
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.
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.
I would have started off my testing by confirming communications between
two IAX modems, I presume you are using HylaFAX too.
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.
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.
I progressed to Asterisk 11 where the T.38 gateway functionality is
better along with other improvements.
What is the output on your system for:
fax show version
Cheers,
Larry.
On 15/11/2016 8:09 PM, tux john wrote:> 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?
> On 13/11/2016, 07:42 Larry Moore <lmoore at starwon.com.au> wrote:
>
> Some additional information which may help you with your installation.
>
> I have 4 IAX Modems named iaxmodem0 - iaxmodem3. I use iaxmodem3
> for outbound fax transmissions.
>
> I created a queue for the other 3 modems, here is my entry in
> queues.conf:
>
> [hylafax-iax]
> strategy=linear
> ringinuse=yes
> autopause=no
> retry=4
> timeout=5
> timeoutpriority=conf
> reportholdtime=no
> joinempty=strict
> leavewhenempty=strict
> musicclass=silence
>
> member => IAX2/iaxmodem2
> member => IAX2/iaxmodem1
> member => IAX2/iaxmodem0
>
> 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:
>
> [silence]
> mode=files
> directory=/usr/local/share/asterisk/silence
> ; ls /usr/local/share/asterisk/silence
> ; 10.gsm
> ;
> ; The file 10.gsm came from
> /usr/local/share/asterisk/sounds/en/silence
>
> 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'.
>
> I've included the console output when an incoming fax is received:
>
> == Using SIP RTP TOS bits 184
> -- Executing [fax at from-itsp:1]
NoOp("SIP/itsp-00000044",
> "Fax Detected 2016-11-13 12:33:40 +0800") in new stack
> -- Executing [fax at from-itsp:2]
GotoIf("SIP/itsp-00000044",
> "0?3:8") in new stack
> -- Goto (from-itsp,fax,8)
> -- Executing [fax at from-itsp:8]
NoOp("SIP/itsp-00000044",
> "Finish if_from-itsp_237") in new stack
> -- Executing [fax at from-itsp:9]
GotoIf("SIP/itsp-00000044",
> "0?10:13") in new stack
> -- Goto (from-itsp,fax,13)
> -- Executing [fax at from-itsp:13]
NoOp("SIP/itsp-00000044",
> "Finish if_from-itsp_238") in new stack
> -- Executing [fax at from-itsp:14]
Set("SIP/itsp-00000044",
> "FAXOPT(gateway)=yes") in new stack
> -- Executing [fax at from-itsp:15]
Queue("SIP/itsp-00000044",
> "hylafax-iax,dRt,,,15") in new stack
> -- Started music on hold, class 'silence', on
> SIP/itsp-00000044
> -- Call accepted by 127.0.0.1 (format alaw)
> -- Format for call is (alaw)
> -- IAX2/iaxmodem2-3086 is ringing
> -- Stopped music on hold on SIP/itsp-00000044
> -- IAX2/iaxmodem2-3086 answered SIP/itsp-00000044
> > 0x89bac000 -- Probation passed - setting RTP source
> address to <ITSP IP Address>:18998
> == Using UDPTL TOS bits 184
> -- Executing [h at from-itsp:1]
GotoIf("SIP/itsp-00000044",
> "0?2:3") in new stack
> -- Goto (from-itsp,h,3)
> -- Executing [h at from-itsp:3]
NoOp("SIP/itsp-00000044",
> "Finish if_from-itsp_239") in new stack
> -- Executing [h at from-itsp:4]
NoOp("SIP/itsp-00000044",
> "Call/Fax Ended 2016-11-13 12:36:41 +0800") in new stack
> -- Hungup 'IAX2/iaxmodem2-3086'
> == Spawn extension (from-itsp, fax, 15) exited non-zero on
> 'SIP/itsp-00000044'
>
> 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
>
> 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.
>
> This is how I performed the majority of my tests.
>
> 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:
>
> context from-itsp {
>
> s => {
> NoOp(Call Received ${STRFTIME(,,%F %T %z)});
> Set(CHANNEL(language)=en_AU);
> Set(DIALTIMEOUT=30);
> Progress();
> NoOp(Call Received from ${CALLERID(name)},
> Tel: ${CALLERID(num)});
> .
> . other conditions checked and extensions dialled
> .
> };
>
> fax => {
> NoOp(Fax Detected ${STRFTIME(,,%F %T %z)});
> Set(FAXOPT(gateway)=yes);
> Queue(hylafax-iax,dRt,,,15);
>
> Set(FAXOPT(gateway)=no);
> &fax-receive(<TSID>,<Header>,FaxMaster,lmoore);
> Hangup();
> };
>
> h => {
> if ( "X${FAXRXFILE}" != "X" )
> {
> &email_rxfax();
> }
> NoOp(Call/Fax Ended ${STRFTIME(,,%F %T %z)});
> };
> };
>
> macro fax-receive( fax-number, header-info, sender, recipient ) {
> /*
> ${ARG1} is Receiving Station Fax Number
> ${ARG2} is Fax Header Information
> ${ARG3} is Fax Sender E-mail Address
> ${ARG4} is Fax Recipient E-mail Address
> */
> NoOp(**** FAX RECEIVE ****);
> Set(FAXOPT(localstationid)=${LOCAL(fax-number)});
> Set(FAXOPT(headerinfo)=${LOCAL(header-info)});
> Set(FROMADDR=${LOCAL(sender)});
> Set(TOADDR=${LOCAL(recipient)});
> NoOp(**** SETTING FAXOPT ****);
> NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)});
> NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)});
> NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)});
> Set(RXSTART=${EPOCH});
> Set(FAXRXPATH=/var/spool/asterisk/fax/received);
> Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});
> NoOp(**** RECEIVING FAX : ${FAXRXFILE} ****);
> ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f);
> NoOp(**** Subroutine Return ****);
> return;
> };
>
> Cheers,
>
> Larry.
>
>
> On 13/11/2016 8:07 AM, Larry Moore wrote:
>
> Is your network/firewall configuration permitting the ports
> for UDPTL, runn the command: udptl show config
>
> UDPTL Global options
> --------------------
> udptlstart: 4000
> udptlend: 4999
> udptlfecentries: 3
> udptlfecspan: 3
> use_even_ports: No
> udptlchecksums: Yes
>
> In your sip configuration for your 'mytrunk' peer have you
set
> applicable options e.g.:
>
> t38pt_udptl=yes,redundancy,maxdatagram=400
>
> 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:
>
> Set(FAXOPT(gateway)=yes)
>
> 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.
>
> Cheers,
>
> Larry.
>
> On 12/11/2016 5:24 AM, tux john wrote:
>
> 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
> inhttp://the-asterisk-book.com/1.6/faxserver.html.
> i do live in Greece and the number is 00302112152130
> the problem is that i am getting the following error and i
> am stuck:
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Executing [00302112152130 at fax-in:1]
> Dial("SIP/mytrunk-00000001",
"IAX2/iaxmodem") in new stack
> -- Called IAX2/iaxmodem
> -- Hungup 'IAX2/iaxmodem-3818'
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Auto fallthrough, channel 'SIP/mytrunk-00000001'
> status is 'CHANUNAVAIL'
> RasPBX*CLI>
> the extensions.conf has
> [fax-in]
> exten => 00302112152130,1,Dial(IAX2/iaxmodem)
> any ideas, please?
>
>
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com
> <http://www.api-digital.com> -- Check out the new Asterisk
> community forum at:https://community.asterisk.org/
> <https://community.asterisk.org/> New to Asterisk? Start
> here:https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> <https://wiki.asterisk.org/wiki/display/AST/Getting+Started>
> asterisk-users mailing list To UNSUBSCRIBE or update options
> visit:http://lists.digium.com/mailman/listinfo/asterisk-users
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
>
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Hi. the fax show version does not work since i am not using the digium modem. the iax2 show peers is the command for me and the output is: PBX*CLI> iax2 show peers Name/Username Host Mask Port Status Description iaxmodem/iaxmod 127.0.0.1 (S) 255.255.255.255 4570 OK (1 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] PBX*CLI> the problem is that in logs i am getting errors and i do not know how to fix it. root at PBX: /var/log/iaxmodem $ more ttyIAX0 [2016-11-16 09:08:12.483144] Registration failed. [2016-11-16 09:13:05.118692] Terminating on signal 15... [2016-11-16 09:21:49.181872] Registration failed. [2016-11-16 09:22:30.731893] Terminating on signal 15... [2016-11-16 09:22:30.759221] Registration failed. [2016-11-16 09:25:11.014642] Registration failed. root at PBX: /var/log/iaxmodem $ Any ideas? On 15/11/2016 5:40 ??, Larry Moore wrote:> 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. > > 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. > > I would have started off my testing by confirming communications > between two IAX modems, I presume you are using HylaFAX too. > > 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. > > 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. > > I progressed to Asterisk 11 where the T.38 gateway functionality is > better along with other improvements. > > What is the output on your system for: > > fax show version > > > Cheers, > > Larry. > > On 15/11/2016 8:09 PM, tux john wrote: >> 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? >> On 13/11/2016, 07:42 Larry Moore <lmoore at starwon.com.au> wrote: >> >> Some additional information which may help you with your >> installation. >> >> I have 4 IAX Modems named iaxmodem0 - iaxmodem3. I use iaxmodem3 >> for outbound fax transmissions. >> >> I created a queue for the other 3 modems, here is my entry in >> queues.conf: >> >> [hylafax-iax] >> strategy=linear >> ringinuse=yes >> autopause=no >> retry=4 >> timeout=5 >> timeoutpriority=conf >> reportholdtime=no >> joinempty=strict >> leavewhenempty=strict >> musicclass=silence >> >> member => IAX2/iaxmodem2 >> member => IAX2/iaxmodem1 >> member => IAX2/iaxmodem0 >> >> 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: >> >> [silence] >> mode=files >> directory=/usr/local/share/asterisk/silence >> ; ls /usr/local/share/asterisk/silence >> ; 10.gsm >> ; >> ; The file 10.gsm came from >> /usr/local/share/asterisk/sounds/en/silence >> >> 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'. >> >> I've included the console output when an incoming fax is received: >> >> == Using SIP RTP TOS bits 184 >> -- Executing [fax at from-itsp:1] NoOp("SIP/itsp-00000044", >> "Fax Detected 2016-11-13 12:33:40 +0800") in new stack >> -- Executing [fax at from-itsp:2] >> GotoIf("SIP/itsp-00000044", "0?3:8") in new stack >> -- Goto (from-itsp,fax,8) >> -- Executing [fax at from-itsp:8] NoOp("SIP/itsp-00000044", >> "Finish if_from-itsp_237") in new stack >> -- Executing [fax at from-itsp:9] >> GotoIf("SIP/itsp-00000044", "0?10:13") in new stack >> -- Goto (from-itsp,fax,13) >> -- Executing [fax at from-itsp:13] NoOp("SIP/itsp-00000044", >> "Finish if_from-itsp_238") in new stack >> -- Executing [fax at from-itsp:14] Set("SIP/itsp-00000044", >> "FAXOPT(gateway)=yes") in new stack >> -- Executing [fax at from-itsp:15] >> Queue("SIP/itsp-00000044", "hylafax-iax,dRt,,,15") in new stack >> -- Started music on hold, class 'silence', on >> SIP/itsp-00000044 >> -- Call accepted by 127.0.0.1 (format alaw) >> -- Format for call is (alaw) >> -- IAX2/iaxmodem2-3086 is ringing >> -- Stopped music on hold on SIP/itsp-00000044 >> -- IAX2/iaxmodem2-3086 answered SIP/itsp-00000044 >> > 0x89bac000 -- Probation passed - setting RTP source >> address to <ITSP IP Address>:18998 >> == Using UDPTL TOS bits 184 >> -- Executing [h at from-itsp:1] GotoIf("SIP/itsp-00000044", >> "0?2:3") in new stack >> -- Goto (from-itsp,h,3) >> -- Executing [h at from-itsp:3] NoOp("SIP/itsp-00000044", >> "Finish if_from-itsp_239") in new stack >> -- Executing [h at from-itsp:4] NoOp("SIP/itsp-00000044", >> "Call/Fax Ended 2016-11-13 12:36:41 +0800") in new stack >> -- Hungup 'IAX2/iaxmodem2-3086' >> == Spawn extension (from-itsp, fax, 15) exited non-zero on >> 'SIP/itsp-00000044' >> >> 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 >> >> 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. >> >> This is how I performed the majority of my tests. >> >> 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: >> >> context from-itsp { >> >> s => { >> NoOp(Call Received ${STRFTIME(,,%F %T %z)}); >> Set(CHANNEL(language)=en_AU); >> Set(DIALTIMEOUT=30); >> Progress(); >> NoOp(Call Received from ${CALLERID(name)}, >> Tel: ${CALLERID(num)}); >> . >> . other conditions checked and extensions dialled >> . >> }; >> >> fax => { >> NoOp(Fax Detected ${STRFTIME(,,%F %T %z)}); >> Set(FAXOPT(gateway)=yes); >> Queue(hylafax-iax,dRt,,,15); >> >> Set(FAXOPT(gateway)=no); >> &fax-receive(<TSID>,<Header>,FaxMaster,lmoore); >> Hangup(); >> }; >> >> h => { >> if ( "X${FAXRXFILE}" != "X" ) >> { >> &email_rxfax(); >> } >> NoOp(Call/Fax Ended ${STRFTIME(,,%F %T %z)}); >> }; >> }; >> >> macro fax-receive( fax-number, header-info, sender, recipient ) { >> /* >> ${ARG1} is Receiving Station Fax Number >> ${ARG2} is Fax Header Information >> ${ARG3} is Fax Sender E-mail Address >> ${ARG4} is Fax Recipient E-mail Address >> */ >> NoOp(**** FAX RECEIVE ****); >> Set(FAXOPT(localstationid)=${LOCAL(fax-number)}); >> Set(FAXOPT(headerinfo)=${LOCAL(header-info)}); >> Set(FROMADDR=${LOCAL(sender)}); >> Set(TOADDR=${LOCAL(recipient)}); >> NoOp(**** SETTING FAXOPT ****); >> NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}); >> NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}); >> NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}); >> Set(RXSTART=${EPOCH}); >> Set(FAXRXPATH=/var/spool/asterisk/fax/received); >> Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID}); >> NoOp(**** RECEIVING FAX : ${FAXRXFILE} ****); >> ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f); >> NoOp(**** Subroutine Return ****); >> return; >> }; >> >> Cheers, >> >> Larry. >> >> >> On 13/11/2016 8:07 AM, Larry Moore wrote: >> >> Is your network/firewall configuration permitting the ports >> for UDPTL, runn the command: udptl show config >> >> UDPTL Global options >> -------------------- >> udptlstart: 4000 >> udptlend: 4999 >> udptlfecentries: 3 >> udptlfecspan: 3 >> use_even_ports: No >> udptlchecksums: Yes >> >> In your sip configuration for your 'mytrunk' peer have you >> set applicable options e.g.: >> >> t38pt_udptl=yes,redundancy,maxdatagram=400 >> >> 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: >> >> Set(FAXOPT(gateway)=yes) >> >> 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. >> >> Cheers, >> >> Larry. >> >> On 12/11/2016 5:24 AM, tux john wrote: >> >> 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 >> inhttp://the-asterisk-book.com/1.6/faxserver.html. >> i do live in Greece and the number is 00302112152130 >> the problem is that i am getting the following error and >> i am stuck: >> == Using SIP RTP TOS bits 184 >> == Using SIP RTP CoS mark 5 >> -- Executing [00302112152130 at fax-in:1] >> Dial("SIP/mytrunk-00000001", "IAX2/iaxmodem") in new stack >> -- Called IAX2/iaxmodem >> -- Hungup 'IAX2/iaxmodem-3818' >> == Everyone is busy/congested at this time (1:0/0/1) >> -- Auto fallthrough, channel 'SIP/mytrunk-00000001' >> status is 'CHANUNAVAIL' >> RasPBX*CLI> >> the extensions.conf has >> [fax-in] >> exten => 00302112152130,1,Dial(IAX2/iaxmodem) >> any ideas, please? >> >> >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided >> by<http://www.api-digital.com>http://www.api-digital.com -- Check >> out the new Asterisk community forum >> at:https://community.asterisk.org/ >> <https://community.asterisk.org/> New to Asterisk? 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