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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Yes, the canreinvite
means Re invite, but there is a consequence in Asterisk configuration.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>For sure, all the
signalisation traffic will go through the asterisk … but for the RTP traffic?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>If canreinvite = No, all
RTP traffic will go through the Asterisk (useful for NATed phoned without
ALG/STUN/…)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>If canreinvite = Yes, the
phones will try to exchange RTP packets directly.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Do you thing there is a
way to allow Re Invite (because you’re right) without the RTP consequence?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Thanks a lot for your
help,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'>Thomas<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-GB
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>De :</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>De la part de</span></b> Rajnish Jain<br>
<b><span style='font-weight:bold'>Envoyé :</span></b> lundi, 19. février
2007 16:25<br>
<b><span style='font-weight:bold'>À :</span></b> Asterisk Users Mailing
List - Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Objet :</span></b> Re: [asterisk-users]
Fax with T.38</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>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.<o:p></o:p></span></font></p>
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<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
<o:p></o:p></span></font></p>
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<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 2/19/07, <b><span style='font-weight:bold'>Thomas
Deillon</span></b> <<a href="mailto:Thomas.Deillon@smart-telecom.ch">Thomas.Deillon@smart-telecom.ch</a>>
wrote:</span></font></span> <o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Hi all,<br>
<br>
I make others tests.<br>
Analog Fax 1 -> PATTON M-ATA -> Asterisk -> PATTON M-ATA -> Analog
Fax2 <br>
<br>
It works only if I use canreinvite= yes.<br>
But all my clients are behind a Nat without ALG or stun stuffs...<br>
<br>
Do you know if canreinvite= yes it's the only way to make it works??<br>
<br>
Thanks a lot for your help, <br>
<br>
Thomas<br>
<br>
<br>
<br>
-----Message d'origine-----<br>
De: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">
asterisk-users-bounces@lists.digium.com</a>] De la part de Thomas Deillon<br>
Envoyé: jeudi, 15. février 2007 11:26<br>
À: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Objet: [asterisk-users] Fax with T.38<br>
<br>
Hi all,<br>
<br>
I make mistakes in my explanation, so I will try to re-explain my problem…<br>
<br>
I want to send fax with FoIP.<br>
Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→
Analog Fax 2 <br>
<br>
In the Patton SN4960 configuration I have :<br>
profile voip FOIP<br>
codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression<br>
codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression<br>
codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression <br>
dtmf-relay signaling<br>
dejitter-max-delay 100<br>
fax transmission 1 relay t38-udp<br>
fax redundancy low-speed 2 high-speed 1<br>
fax detection fax-frames<br>
modem transmission 1 bypass g711alaw64k<br>
modem bypass-method nse <br>
<br>
On Patton M-ATA :<br>
1. codec alaw<br>
2. codec ulaw<br>
3. codec g729<br>
No silence suppression on these codecs.<br>
I not use this option "FAX without T.38(Use G.711 fax)"<br>
<br>
<br>
On asterisk side I have:<br>
[general] <br>
context=default<br>
bindport=5060<br>
bindaddr=<a href="http://0.0.0.0">0.0.0.0</a> <br>
srvlookup=yes<br>
disallow=all<br>
allow=alaw<br>
dtmfmode = rfc2833<br>
rtcachefriends=yes<br>
realm=vtxvoip<br>
useragent=VTX SIP<br>
rtupdate=yes <br>
language=en<br>
tos=184<br>
notifyringing=yes<br>
t38pt_udptl=yes<br>
<br>
And t38pt_udptl=yes in the 2 PATTONs sip accounts …<br>
<br>
<br>
Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 ….<br>
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:<br>
<br>
[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) <br>
[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a
codec translation path from alaw to g729<br>
[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a
codec translation path from alaw to g729 <br>
<br>
<br>
What I really not understand it's why asterisk try to translate from ulaw to
g729 !!!<br>
I disallow all and allow just the alaw codec … more than this, I remove the
g729 licence file …<br>
<br>
Do you have an idea for me ?? <br>
<br>
Thanks a lot,<br>
<br>
Thomas<br>
<br>
<br>
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