[asterisk-users] Fax with T.38
Thomas Deillon
Thomas.Deillon at smart-telecom.ch
Mon Feb 19 07:49:20 MST 2007
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] 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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