Trixbox - Fax Server
Submitted by JKbutchoy on Wed, 03/14/2007 - 2:02am.
Trixbox can handle incoming fax & send it to the specified email address, well with some tweeking & reading I was able to set trixbox act as a fax server. Callers can choose whom to send the fax. The caller is given an options to make a voice call or a fax sending. Fax receipient can be an extension user or a virtual number then forwarded to his respective email address. It works well with digium card 4port fxo.
Re: Trixbox - Fax Server
Sun, 02/04/2007 - 12:05am
Re: Trixbox - Fax Server
Wed, 02/07/2007 - 2:13pm
Re: Trixbox - Fax Server
Tue, 03/27/2007 - 5:37pm
Fax Server
My example will focus to a single extension, from here you can create your own.
Ext 101 - 101email@email.com
1. Create your call flow for the IVR. You must have a sequence on the call flow where an option to send fax.
2. On that IVR, put the ext # (101) on the digits menu then enter custom context to the Custom App (ex. custom-fax101,101,1).
Uncheck Enable Direct Dial on IVR menu
3. Create a custom context in extensions_customs.conf
to make it easy, here is an example:
[custom-fax101]
exten => 101,1,Answer
exten => 101,2,Goto(in_fax1,1)
exten => in_fax1,1,StopPlayTones
exten => in_fax1,2,GotoIf($["${FAX_RX}" = "system"]?3:analog_fax,1)
exten => in_fax1,3,Macro(faxreceive)
exten => in_fax1,4,system(/var/lib/asterisk/bin/fax-process.pl --to 101email@email.com --from ${FAX_RX_FROM} --subject "Fax Received ${CALLERID(number)} ${CALLERID(name)}" --attachment ${CALLERID(number)}fax.pdf --type application/pdf --file ${FAXFILE});
exten => in_fax1,5,Hangup
exten => analog_fax,1,GotoIf($["${FAX_RX}" = "disabled"]?4:2) ;if fax is disabled, just hang up
exten => analog_fax,2,Set(DIAL=${DB(DEVICE/${FAX_RX}/dial)});
exten => analog_fax,3,Dial(${DIAL},20,d)
exten => analog_fax,4,Hangup
*this is my own set-up, you must look-up to the field with my defined extension & email address (101 & 101email@email.com)
You can also do some adjustments in sendmail if you have another email server, but by default trixbox will just send the fax in pdf to the specified email. (You must have an internet access)
4. On the General Settings (freepbx), leave blank on the email address to have faxes emailed to & create an email address (mine is faxserver@email.com) to the field faxes appear to come from. This can be non existing email address.
5. On Inbound routes (freepbx), select system to Fax Extension, then your general email add to the Fax email. This is where your common fax messages will be sent, assuming all inbound fax must be forwarded to as emails.
Now, if somebody calls-in & the IVR is set to handle the call, then he has an option to call my extension or send a fax to me. He can also send fax to our general email address.
Keep in mind that if somebody will send you fax using PC/modem/auto fax, this will be automatically routed to the general email for receiving fax.
I hope everything is clear, with just some experiments you can built your own inbound fax server.
maraming salamat trixbox ! ! ! mabuhay ! ! !
Re: Trixbox - Fax Server
Sun, 03/04/2007 - 12:39pm
Thanks very much for this contribution. The fax works like a champ. One thing that I need help with is that the subject line contains some character that I will like to get rid of.
The sample subject line of my fax received is : Fax from 7099557406 bezy%20John%20%20%20%20, the phone number is my number and the name is the caller name. How do I get ride of %.
Re: Trixbox - Fax Server
Wed, 03/14/2007 - 2:02am
Hello,
Salamat po sa post.. I would just like to seek advice about the problem on my installation. I have installed TrixBox v2.0.0 (asterisk-1.2.14) and installed asterfax-1.0-free edition. I can now successfully run the asterfax.sh but my test fax attempts failed. I just got “ERROR Hangup received before FaxTransmittedEvent. Please check that you have patched TxFax” on the asterfax console. Do you have any idea what this means? Btw, I’m using a Welltech fxo as my gateway to my analog pstn line; I’m just not sure if it has effect on it but I can do incoming and outgoing voicecall with problem. Any advice is highly appreciated.
Maraming salamat po..
:-(
Pards,
Im not using
Mon, 03/19/2007 - 11:16pm
Thank You!
Wed, 03/28/2007 - 8:29am
JKbutchoy, the fax from field is captured correctly except for the fact that the spaces between each letter or number is filled with %%%% (percentage sign). I have tried to get rid of this annoying sign but no luck. The fax server works like a champ. Thans for your contributions.
My fax is setup on a sip
Thu, 03/29/2007 - 11:01am
My fax is setup on a sip trunk. I have tried to play with the CID handling. I am not using a digium or any pci card.
Below is the exact copy of the subject line when a fax is received:
Subject: Fax from 212444xxxx Igocome%20%20%20%20%20%20%20SDavid.
The %20 makes it difficult to search for a fax message. It is also very annoying.
Any helpful input to resolve this will be highly appreciated.
You're welcome, I'm not sure
Thu, 03/29/2007 - 9:57pm
You're welcome, I'm not sure what to do with the 20% sign on the Subject. Maybe its the actual savings you get (20% off) . . . . just kiding. for that I will just charge you $1.50 a month.
What is your current set-up; You are using it as your IPPBX with Fax Service or just to handle incoming fax?
If you recieve the call on another sip extension, does the CID displays properly?
My current setup is IPPBX
Fri, 03/30/2007 - 2:25am
My current setup is IPPBX WITH fax service. I still receive telephone calls on the same system. When fax comes the system auto detect fax and received it without disturbing or interfering with my phone. Yes, the CID display properly on all other sip extensions. I believe the system auto fills the gap/space between each word in the subject line with the 20% off. Thanks for your help.
How succesful is Trixbox-Fax over VoIP???
Tue, 04/10/2007 - 3:56am
Hello, all!
Looks like some of you may have the experience I'm looking for!
I'm going to get a Asterisk set up soon on a remote VPS somewhere (hence NO PSTN on it!), and I've been trying to figure out how well Asterisk/Asterfax/etc can get faxes to/from voip trunks?
Searching the web, I only get conflicting opinions, which tend toward pessimism....
Has anybody done it over international trunks? Over *satellite* international voip trunks with jitter showing on the ping's????
How is T38 coming along on Asterisk?
Thank for any opinions/advice.
Chris
how to remove those %20
Wed, 04/11/2007 - 12:38pm
Edit /var/lib/asterisk/bin/fax-process.pl
goto line 55
} elsif ($cmd eq "--type") {
and insert the following line above it
$subject =~ s/\%20/ /g
So it should now look like
} else {
# It's a single word
$subject = $tmp;
}
$subject =~ s/\%20/ /g
} elsif ($cmd eq "--type") {
my $tmp = shift @ARGV;
That searches and replaces the %20 with a space which is what %20 translates to.
Thanks fskrotzki, edith my
Sat, 04/14/2007 - 2:01am
re:Thanks fskrotzki, edith my
Mon, 04/16/2007 - 9:38am
Something is typed wrong. If you run the command without anything does it complain about a error?
/var/lib/asterisk/bin/fax-process.pl
Should give you a line showing the command line options. If not something is typed wrong.
I'm going to update my post to include the other common characters that are escaped in a second.
update: how to remove those %2x where x is [0-9,a-f]
Tue, 04/24/2007 - 10:11am
edit the /var/lib/asterisk/bin/fax-process file and at line 55 change it to look like the following:
BEFORE:
} else {
# It's a single word
$subject = $tmp;
}
} elsif ($cmd eq "--type") {
my $tmp = shift @ARGV;
$ct = $tmp if (defined $tmp);
AFTER:
} else {
# It's a single word
$subject = $tmp;
}
$subject =~ s/\%20/ /g;
$subject =~ s/\%21/\!/g;
$subject =~ s/\%22/\"/g;
$subject =~ s/\%23/\#/g;
$subject =~ s/\%24/\$/g;
$subject =~ s/\%25/\%/g;
$subject =~ s/\%26/\&/g;
$subject =~ s/\%27/\'/g;
$subject =~ s/\%28/\(/g;
$subject =~ s/\%29/\)/g;
$subject =~ s/\%2a/\*/g;
$subject =~ s/\%2A/\*/g;
$subject =~ s/\%2b/\+/g;
$subject =~ s/\%2B/\+/g;
$subject =~ s/\%2c/\,/g;
$subject =~ s/\%2C\,/g;
$subject =~ s/\%2d/\-/g;
$subject =~ s/\%2D\-/g;
$subject =~ s/\%2e/\./g;
$subject =~ s/\%2E\./g;
$subject =~ s/\%2f/\//g;
$subject =~ s/\%2F\//g;
} elsif ($cmd eq "--type") {
my $tmp = shift @ARGV;
$ct = $tmp if (defined $tmp);
This change is valid for asterisk 1.2.12 to 1.2.17-1
I've not tested or played with the 1.4.x set yet. And yes there is probably a cleaner way to do this in a loop but I can read perl and not really program in it, but this works so I'm happy.
Also if you pull down the freeepbx 2.2.1 update it has the first fix installed correctly.
Custom context generalized?
Wed, 05/09/2007 - 6:00pm
It seems we should be able to generalize this custom context.
Is the following right?
On that IVR, put the ext # (101) on the digits menu then enter custom context to the Custom App (ex. custom-fax,101,1).
3. Create a custom context in extensions_customs.conf
to make it easy, here is an example:
[custom-fax]
exten => $1,1,Answer
exten => $1,2,Goto(in_fax1,1)
.
.
.
?
If you generalize the
Tue, 05/15/2007 - 3:23am
If you generalize the context, then the fax message cannot be sent to the specific email address or recepient. With this my so called fax server, you can define the user's number & his email address. ext 101 to email101@mail.com, ext 102 to email102@gmail.com . . .etc.
I hope Trixbox can include this feature so that inputs, edit & delete will be easier.
I have successfully activated around 300 users.
(,")
NO GO - Help Please
Wed, 05/23/2007 - 2:57pm
Hi,
I followed all the instructions on configuring the fax here, but still NO GO on my trixbox server. I'm using version 2.2 but nothing happens when I send the fax. I call to my line (conected via ZAP), I dial the extension nº, I receive fax tone, but nothing happens after this. I'm sending over the log here. Please any help will be appreciated.
Regards,
E.
May 23 17:49:22 DEBUG[21125] chan_zap.c: DTMF digit: 3 on Zap/1-1
May 23 17:49:22 DEBUG[21125] channel.c: Scheduling timer at 0 sample intervals
May 23 17:49:22 DEBUG[21125] pbx.c: Oooh, got something to jump out with ('3')!
May 23 17:49:22 DEBUG[21125] chan_zap.c: DTMF digit: 0 on Zap/1-1
May 23 17:49:22 DEBUG[21125] chan_zap.c: DTMF digit: 3 on Zap/1-1
May 23 17:49:23 DEBUG[21125] chan_zap.c: DTMF digit: 0 on Zap/1-1
May 23 17:49:23 DEBUG[21125] app_db.c: Ignoring, no parameters
May 23 17:49:23 DEBUG[21125] pbx.c: Expression result is '1'
May 23 17:49:23 DEBUG[21125] app_macro.c: Executed application: Set
May 23 17:49:23 DEBUG[21125] app_macro.c: Executed application: Set
May 23 17:49:28 DEBUG[21125] chan_zap.c: DTMF digit: f on Zap/1-1
May 23 17:50:01 DEBUG[21125] app_rxfax.c: ==============================================================================
May 23 17:50:01 DEBUG[21125] app_rxfax.c: Fax receive not successful - result (3) Timed out waiting for the first message.
May 23 17:50:01 DEBUG[21125] app_rxfax.c: ==============================================================================
May 23 17:50:01 DEBUG[21125] app_macro.c: Executed application: rxfax
May 23 17:50:01 DEBUG[21125] pbx.c: Function result is ''
May 23 17:50:01 DEBUG[21125] pbx.c: Function result is ''
May 23 17:50:01 DEBUG[21125] pbx.c: Function result is ''
May 23 17:50:01 DEBUG[21125] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
May 23 17:50:01 DEBUG[21125] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2007-05-23 17:49:14','','','in_fax1','custom-fax3030', 'Zap/1-1','','Hangup','',47,47,'ANSWERED',3,'','1179956949.322')
May 23 17:50:01 DEBUG[21125] chan_zap.c: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1
May 23 17:50:01 DEBUG[21125] chan_zap.c: disabled echo cancellation on channel 1
May 23 17:50:01 DEBUG[21125] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
May 23 17:50:01 DEBUG[21125] chan_zap.c: Updated conferencing on 1, with 0 conference users
You get fax tone then send
Wed, 05/23/2007 - 6:21pm
I get tone, but the fax does not start to pass
Thu, 05/24/2007 - 5:20am
Hi JK,
What happens is the following, I call to the line conected to the TB, the IVR answers, I dial the Fax extension, it gives fax tone, I press send on the fax but it simply stays there, the fax does not start to transmit. On the fax status monitor it stays in conecting status. The log for the TB, is the one I posted before, on the maillog log, I dont see any log for my fax extension yesterday (which is the day I started working on this). In advance, thanks for your help and for this post, it looks as a really cool solution.
Ps: The TB can send email, because the users are getting their voicemail every day.
THX
Can you post one of the
Tue, 05/29/2007 - 5:52pm
Can you post one of the custom context? It should looks like this as an example for extension 101:
[custom-fax101]
exten =>101,1,Answer
exten => 101,2,Goto(in_fax101,1)
exten => in_fax101,1,StopPlayTones
exten => in_fax101,2,GotoIf($["${FAX_RX}" = "system"]?3:analog_fax,1)
exten => in_fax101,3,Macro(faxreceive)
exten => in_fax101,4,system(/var/lib/asterisk/bin/fax-process.pl --to user101@email.com --from ${FAX_RX_FROM} --subject "Fax Received ${CALLERID(number)} ${CALLERID(name)}" --attachment ${CALLERID(number)}fax.pdf --type application/pdf --file ${FAXFILE});
exten => in_fax101,5,Hangup
exten => analog_fax,1,GotoIf($["${FAX_RX}" = "disabled"]?4:2) ;if fax is disabled, just hang up
exten => analog_fax,2,Set(DIAL=${DB(DEVICE/${FAX_RX}/dial)});
exten => analog_fax,3,Dial(${DIAL},20,d)
exten => analog_fax,4,Hangup
I do not know how you setup
Wed, 05/30/2007 - 5:35am
I do not know how you setup your system but with mine there is no need to dial the fax extension, it detect fax tone as soon fax comes in. But now, I have a dedicated fax number, what I did is created an ivr with the just the fax extenstion without any announcement. It works like a cham. Thanks to JKbutchoy!!!
using one telephone line for phone & fax
Tue, 07/17/2007 - 9:20am
NAT to NAT SIP Solution
Sat, 09/08/2007 - 8:17pm
To ALL Trixboxers,
I have decided to give back to the community who has given so much. SIP NAT to NAT has been a thorn in all our sides for quite some time. There has been a lot of discussion whether it works or doesn't work and many types of work around solutions have been presented.
After many attempts, I am pretty sure that I have found a solution that solves the NAT to NAT problem and it works reliably for me. I have not seen anyone outline in the solution to the NAT to NAT resolution. So EDIT those files, CONFIGURE that router, Turn on that FIREWALL, here is what you need to do.
There are three files to edit in /etc/asterisk are
file rtp.conf
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Grandstreams used to be stuck using 10000, this is now fixed.
; Webmin uses udp/10000, so we avoid that.
rtpstart=8000
rtpend=20000
----------------------------------------------------------------
file sip.conf
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
externip = public IP address of ROUTER
localnet=local IP address of Trixbox/255.255.255.0
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf
------------------------------------------------------------------
file sip_nat.conf
externip = public IP address of Router
localnet=local IP address of Trixbox/255.255.255.0
externrefresh=10
nat=yes
Now, you must also configure your router to perform port forwarding on the following port ranges.
4560 to 4580 this is for IAX protocol
21 to 23 this is for ftp and ssh
8000 to 8079 this is to leave 8080 open
8079 to 9999 this is to leave 10000 open for webmin
10001 to 20000 this is for the remaing rtp
5060 to 6000 these ports are for sip
Now you can tighten up these port forward ranges if necessary but I find they work fine for the systems that I have installed. I have these solutions in networks with VPN routers being used as voice solutions.
I hope that these instructions are helpful and finally put to rest at least for awhile the problems that SIP with NAT to NAT can work reliably.
If anyone has any questions, they can reply here. I also do consulting during my free time if any assistance is required.
Good luck with the instructions, they should help.
Sincerely,
Richard Mataka
Trixbox Fax server
Thu, 06/26/2008 - 7:53am
I am using Trixbox 2.6.0.0. I have an analog dedicated fax line coming into the system. When I direct it to a real fax machine, it almost always works. But, it almost never works when I send it to the "system" fax option. (Are there any options anyone knows of that should be altered? What about echo cancellation?)
With that said, I also have a dedicated SIP phone number for incoming faxes. When I send that to the "system" fax option (NVfax), it works about 50% of the time.
The load on our system is rarely above 1%, so there is plenty of processing power, memory (never need to use the swap file), and disk space. This makes very little sense to me, as I would expect the fax line to be far more reliable than a SIP call coming in over our cable modem service.
Are there any parameters/config files that anyone has altered from the standard with Trixbox that have increased the probability of successful receipt of faxes?
fax server from fresh install
Sat, 09/13/2008 - 7:04am
before trying jkbutchoy's instructions, test your trixbox's fax server functionality first. it should work from fresh install.
go to setup -> feature codes and make sure that dial system fax is enabled (default setting is 666|checked|enabled);
test it by dialing 666 from a any extension and you should hear a fax tone;
if at this point, you don't get a fax tone, then something is wrong...
if it works, then you must setup sendmail or postfix to handle mail sending. please refer to mail setup thread.
the easiest way of receiving fax is by simply transferring incoming calls to 666. trixbox will then send the received fax to the email address defined in general settings.
some people (in my country) prefer to make voice calls then ask "fax tone please?" rather than let the fax machines talk to each other. most don't listen to the ivr and just wait until someone answers.
nevertheless, i think jkbutchoy's solution is the easiest if you want to automate incoming fax. mabuhay ka kabayan!
Salamat Bro,
Guys, Any
Thu, 09/18/2008 - 1:07am
Salamat Bro,
Guys, Any donation for my post? Just kidding. . .
Individual Fax Extension
Fri, 10/17/2008 - 1:11pm
Read through this quite a bit, but i am not seeing exactly what i'm looking for, but because of this thread, i was able to send a fax to 666 via transfer and forward. but initial fax detection (CED?) would be great. All users have a DID.
Here is a thread that i posted just a little while ago, http://trixbox.org/forums/trixbox-ce-development-forum/fax-develo...
Every user needs to have their own line that has the same DID for fax, just like their current setup.
Any help would be awesome.
Individual Fax (DID) ?
Wed, 10/22/2008 - 2:42am
First, you must find a way to connect your Trixbox to the DID line. If you are using Digium FXO Card or any FXO Card, you cannot connect it to the DID lines. My solution applies not only to analog lines but even on SIP trunks. The Fax Server function here follows the same process as an incoming voice call. Callers are guided thru by the IVR to reach the desired destination.
Member Since:
2006-11-13