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 <[email protected]> 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@from-itsp:1] NoOp("SIP/itsp-00000044",
"Fax Detected 2016-11-13 12:33:40 +0800") in new stack
-- Executing [fax@from-itsp:2]
GotoIf("SIP/itsp-00000044", "0?3:8") in new stack
-- Goto (from-itsp,fax,8)
-- Executing [fax@from-itsp:8] NoOp("SIP/itsp-00000044",
"Finish if_from-itsp_237") in new stack
-- Executing [fax@from-itsp:9]
GotoIf("SIP/itsp-00000044", "0?10:13") in new stack
-- Goto (from-itsp,fax,13)
-- Executing [fax@from-itsp:13] NoOp("SIP/itsp-00000044",
"Finish if_from-itsp_238") in new stack
-- Executing [fax@from-itsp:14] Set("SIP/itsp-00000044",
"FAXOPT(gateway)=yes") in new stack
-- Executing [fax@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@from-itsp:1] GotoIf("SIP/itsp-00000044",
"0?2:3") in new stack
-- Goto (from-itsp,h,3)
-- Executing [h@from-itsp:3] NoOp("SIP/itsp-00000044",
"Finish if_from-itsp_239") in new stack
-- Executing [h@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@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?
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