[asterisk-users] Fax with T.38

[Thomas Deillon]

Hi all,

 

I install the last version of Asterisk and I tried to send faxes, but
nothing works.

Here is my configuration:

 

Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA
<----> Analog Fax 2

 

I tried Analog Fax 2 -> Analog Fax but nothing works!!

 

In the Patton configuration I put G711 and no silence suppression.

 

In asterisk I have some errors :

[Feb 14 11:28:55] WARNING[10547]: channel.c:3033
ast_channel_make_compatible: No path to translate from
SIP/sip_trunk_gva-mg-02-006f37f0(256) to SIP/0625037998-006de430(8)

 [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to
find a codec translation path from alaw to g729

[Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to
find a codec translation path from alaw to g729

 [Feb 14 11:29:05] NOTICE[10547]: rtp.c:772 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible. Client IP: xx.xx.xx.xx

 

In my SIP.conf file:

 

[general]

context=default         ; Default context for incoming calls

bindport=5060           ; UDP Port to bind to (SIP standard port is
5060)

bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
to all)

srvlookup=yes           ; Enable DNS SRV lookups on outbound calls

disallow=all                    ; First disallow all codecs

allow=g729

allow=gsm

allow=alaw                      ; Allow codecs in order of preference

dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF.
Default: rfc2833

rtcachefriends=yes

realm=vtxvoip

useragent=VTX SIP

rtupdate=yes

language=en

tos=184

notifyringing=yes

t38pt_udptl=yes

 

I have the definition of the phone in DB.

voip-test-01*CLI> sip show peer 0625037998

voip-test-01*CLI>

 

  * Name       : 0625037998

  Realtime peer: No

  Secret       : <Set>

  MD5Secret    : <Not set>

  Context      : sipresidential

  Subscr.Cont. : <Not set>

  Language     : fr

  AMA flags    : Unknown

  Transfer mode: open

  CallingPres  : Presentation Allowed, Not Screened

  Callgroup    :

  Pickupgroup  :

  Mailbox      : 0625037998 at default

  VM Extension : asterisk

  LastMsgsSent : 0/0

  Call limit   : 6

  Dynamic      : Yes

  Callerid     : "0625037998" <0625037998>

  MaxCallBR    : 384 kbps

  Expire       : -1

  Insecure     : no

  Nat          : RFC3581

  ACL          : No

  T38 pt UDPTL : Yes

  CanReinvite  : No

  PromiscRedir : No

  User=Phone   : No

  Video Support: No

  Trust RPID   : No

  Send RPID    : No

  Subscriptions: Yes

  Overlap dial : Yes

  DTMFmode     : inband

  LastMsg      : 0

  ToHost       :

  Addr->IP     : (Unspecified) Port 0

  Defaddr->IP  : 0.0.0.0 Port 5060

  Reg. exten   :

  Def. Username: 0625037998

  SIP Options  : (none)

  Codecs       : 0xc (ulaw|alaw)

  Codec Order  : (alaw:20,ulaw:20)

  Auto-Framing:  No

  Status       : UNKNOWN

  Useragent    : Patton Smartlink MATA <4.01.001 OE EN MA
(0412)><00a0ba01a154>

  Reg. Contact : sip:0625037998 at xx.xx.xx.xx:5060

 

Thanks a lot for your help,

 

Thomas

 

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