[asterisk-users] Fax with T.38

[Thomas Deillon]

Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration.

For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic?

If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…)

If canreinvite = Yes, the phones will try to exchange RTP packets directly.

 

Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence?

 

Thanks a lot for your help,

 

Thomas

 

________________________________

De : asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] De la part de Rajnish Jain
Envoyé : lundi, 19. février 2007 16:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Fax with T.38

 

A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones.  The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work.

 


 

On 2/19/07, Thomas Deillon <Thomas.Deillon at smart-telecom.ch> wrote: 

Hi all,

I make others tests.
Analog Fax 1 -> PATTON M-ATA -> Asterisk -> PATTON M-ATA -> Analog Fax2 

It works only if I use canreinvite= yes.
But all my clients are behind a Nat without ALG or stun stuffs...

Do you know if canreinvite= yes it's the only way to make it works??

Thanks a lot for your help, 

Thomas



-----Message d'origine-----
De: asterisk-users-bounces at lists.digium.com [mailto: asterisk-users-bounces at lists.digium.com <mailto:asterisk-users-bounces at lists.digium.com> ] De la part de Thomas Deillon
Envoyé: jeudi, 15. février 2007 11:26
À: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: [asterisk-users] Fax with T.38

Hi all,

I make mistakes in my explanation, so I will try to re-explain my problem…

I want to send fax with FoIP.
Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 

In the Patton SN4960 configuration I have :
profile voip FOIP
codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression
codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression
codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression 
dtmf-relay signaling
dejitter-max-delay 100
fax transmission 1 relay t38-udp
fax redundancy low-speed 2 high-speed 1
fax detection fax-frames
modem transmission 1 bypass g711alaw64k
modem bypass-method nse 

On Patton M-ATA :
1. codec alaw
2. codec ulaw
3. codec g729
No silence suppression on these codecs.
I not use this option "FAX without T.38(Use G.711 fax)"


On asterisk side I have:
[general] 
context=default
bindport=5060
bindaddr=0.0.0.0 
srvlookup=yes
disallow=all
allow=alaw
dtmfmode = rfc2833
rtcachefriends=yes
realm=vtxvoip
useragent=VTX SIP
rtupdate=yes 
language=en
tos=184
notifyringing=yes
t38pt_udptl=yes

And t38pt_udptl=yes in the 2 PATTONs sip accounts …


Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 ….
I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS:

[Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) 
[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729
[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 


What I really not understand it's why asterisk try to translate from ulaw to g729 !!!
I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file …

Do you have an idea for me ?? 

Thanks a lot,

Thomas


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