Be aware: T.38 is not T.38, there are still a great many interoperability issues out there!

Version information

• Asterisk 1.2 has no support for T.38.
• Asterisk 1.4 supports only T.38 fax pass through; there is however a third party way using HylaFax and OPAL to send and receive fax through Asterisk 1.4. See also rejected patch 12931 that includes a T.38 gateway. Attractel offers a commercial T.38 gateway solution for Asterisk.
• In Asterisk 1.6 also origination and termination features will be added (with gateway functionality still missing)

FAQ

Q: Does Asterisk support T.38 ATAs behind NAT with canreinvite=no ?
A: See below

Q: Can I terminate T.38 calls to PSTN with Asterisk T.38 passthrough(via Zaptel)?
A: yes, http://bugs.digium.com/view.php?id=12931
CallWeaver supports T.38 termination and gateway operation.

ATA COMPATIBILITY

ATA1(CALLER)	FIRMWARE	NAT	ASTERISK VERSION	ATA2/GW(CALLED)	FIRMWARE	NAT	works?	notes
HT496	1.0.3.44	yes	SVN TRUNK 40000	HT496	1.0.3.44	yes	yes	example
HT496	1.0.3.64	yes	SVN-branch-1.4-r53152	Patton SN4960	T4	no	yes*	sometimes bad quality(ht496 unregister after fax is sent)
SPA2100	3.3.6	no	Branch 1.2+patches	Cisco AS5300	IOS 12.3	no	yes	example
Kapanga	2152b	no	SVN-branch-1.4-r47911	Kapanga	2152b	no	yes*	*receiving kapanga crash
Kapanga	2156	yes	SVN-branch-1.4-r53152	Patton SN4960	T4	no	yes
Patton SN4524	3.20	yes	Asterisk 1.4.1	Patton SN4524	3.20	yes	yes
SPA2102	5.1.6	yes	Asterisk 1.4.2	Patton SN4960	R4.1	no	NO
SPA2102	5.1.1	yes	Asterisk 1.4.2	Patton SN4960	R4.1	no	yes
SPA2102	5.2.5	yes	Asterisk 1.4.19	Gafachi	*	no	yes
Micronet SP5002/S	113	yes	Asterisk 1.4.3	CISCO IOS	N/A	yes	yes
Grandstream HT502	1.0.0.39	yes	Asterisk 1.4.3	Patton SN4960	4.1	yes	yes
Grandstream HT502	1.0.0.39	yes	Asterisk 1.4.4	Grandstream HT502	1.0.0.39	yes	yes
Grandstream GXW4004	1.0.0.39	yes	Asterisk 1.4.4	Grandstream HT502	1.0.0.39	yes	yes
Gafachi UAS	110.05	yes	Asterisk 1.4.6+	Grandstream HT287 (aka HT286 v3.0)	1.1.0.3	no	yes
Grandstream HT287 (aka HT286 v3.0)	1.1.0.3	no	Asterisk 1.4.6+	Gafachi UAS	110.05	yes	yes
Grandstream HT503	Prg.1.0.0.5	yes	Asterisk 1.4.17	Grandstream HT503	Prg.1.0.0.5	yes	yes
Mediatrix 1102	v5.0.19.124	yes	Asterisk 1.4.18	Mediatrix 1102	v5.0.19.124	yes	yes	Same behaviour for all of mediatrix products line

Installation Procedures for Asterisk 1.4 with T.38

Installation Note: By default, Asterisk doesn't set many of the T.38 settings that may be required to interface with third party SIP trunks. You must make these changes within the chan_sip.c file before doing a make install. The T.38 configuration settings is located in chan_sip.c and can be edited with VI or your favorite text editor. The setting is modified with the line that reads as follows:

/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;

To make changes to the T38 configuration, simply add in the changes as described in the sections above that line. For example, if you need T38FAX_VERSION_1 simply edit the file and change the 0 to 1. If you can support 12000 and 14400, simply add it to the end of the line. An example configuration with T38 verison 1 and adding 12000 and 14400 follows:

static int global_t38_capability = T38FAX_VERSION_1 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600 | T38FAX_RATE_12000 | T38FAX_RATE_14400;

Install

 cd /usr/src
 svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 
(or for trunk:
  svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk
)
 cd asterisk-1.4 (cd asterisk-trunk)
 ./configure;make;make install
 asterisk -vvvdc

Config

/etc/asterisk/sip.conf:

 [general]
 ;NEEDED!!!
 t38pt_udptl = yes

 [200] 
 type=friend
 context=from-sip     
 host=dynamic           
 secret=200
 canreinvite=yes
 t38pt_udptl = yes


or take a look at a NATted variant:

 [200] 
 type=friend
 context=from-sip     
 host=dynamic           
 secret=200
 canreinvite=no
 nat=yes
 t38pt_udptl = yes

 [201] 
 type=friend
 context=from-sip
 host=dynamic           
 secret=201
 canreinvite=yes
 t38pt_udptl = yes


/etc/asterisk/extensions.conf:

 [from-sip]
 exten => 200,1,Dial(SIP/${EXTEN}|300)
 exten => 201,1,Dial(SIP/${EXTEN}|300)

Now dial 201 from 200.

Additional Troubleshooting

Depending on your fax device (such as the Linksys 3102) you may have to edit the udptl.conf file. The error correction type that is sent is usually the culprit of many problems with ATAs and T.38 providers.

Also, try using:

 t38_udptl=yes
 t38pt_rtp=no
 t38pt_tcp=no

... in the general section of the sip.conf and under the VoIP provider account as well as the fax account.

Also, crank down the speed of the fax machine to the slowest speed possible. Some ATAs work better like this.

Installation Procedures for Asterisk T.38 Over FreePBX 2.4

1. Setup /etc/asterisk/udptl.conf manually
2. add "t38pt_udptl=yes" to /etc/asterisk/sip_general_custom.conf

Step 2 basically tells asterisk that ALL of your SIP peers are T.38 capable (which is probably not true, but not really harmful either)

HowTo debug & send bugreports

Please read http://www.digium.com/bugguidelines.html

1) Prepare test environment (reduce the ammount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:

  console => notice,warning,error,debug

3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:

  set debug 4 (latest trunk "core debug 4")
  set verbose 4 (latest trunk "core verbose 4")
  sip debug

5) Save complete console log to file
6) Create bugreport at http://bugs.digium.com
7) Attach saved file to the bug.

See also

• Asterisk fax
• T.38
• T38modem configuration with Asterisk
• Zoiper: SIP /IAX softphone with T.38 support incl. print2fax driver