Could anybody give me an authoritative answer on whether Asterisk can
support T.38 pass-through when the clients are behind NAT? We have
Asterisk servicing clients behind NAT (with nat=route, canreinvite=no)
and would love to get T.38 going but have had no luck so far. The
following case:
http://bugs.digium.com/view.php?id=7844
...suggests that T.38 *does* now work for clients behind NAT but I have
the latest SVN trunk but still cannot get it to work? On the other side
I have seen on this list only 2 weeks or so ago:
http://www.mail-archive.com/[email protected]/msg172556.html
This suggests that T.38 does *NOT* work behind NAT? So, can anybody
save me the trouble and tell me how it is. Am I on a hiding to nothing
trying to get T.38 going with NAT? Please put me out of my misery! :)
Cheers,
Ray
PS. Does anybody know whether OpenPBX would support T.38 and NAT
configurations? This was my backup plan if I couldn't get it to go in
Asterisk.
Thomas Deillon wrote:
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 :* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *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* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> 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: [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> [mailto:
[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>] 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 <http://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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