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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Jeremy… Just curious, do you use SIP/RTP at all for fax/modems
?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Jeremy
Mann<br>
<b>Sent:</b> Wednesday, May 28, 2008 9:47 AM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] TE2XXP and modems<o:p></o:p></span></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;color:#1F497D'>Just make sure
echocancel is disabled when bridged and you should be fine. Also whatever
card you get, make sure the timing is set properly. I have a 2-port
sangoma much like your setup, the PRI is the main timing source, the 2<sup>nd</sup>
port goes to a Nortel which inherits timing from the PRI as well. Once
timing was setup properly my modems/faxes work fine.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;color:#1F497D'><o:p> </o:p></span></p>
<div>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Joe
Carroll<br>
<b>Sent:</b> Wednesday, May 28, 2008 8:31 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] TE2XXP and modems<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>We are doing something similar today with successful
phones, fax, and modem… PURE TDM.. no SIP<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Example Below….<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal>PSTN ----<PRI(s)> ---- <Asterisk>
---------------- <Nortel PBX> ---- Phones, faxes or modems<span
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>
|<o:p></o:p></span></p>
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font-family:"Calibri","sans-serif";color:#1F497D'>
|<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>
|<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>
|</span>------------------
<Avaya PBX>---- Phones, faxes or modems<span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Olivier<br>
<b>Sent:</b> Wednesday, May 28, 2008 2:36 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] TE2XXP and modems<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>In fact, today's setup is :<br>
PSTN ----<PRI> ---- Alcatel OXE PBX ---- Phones, faxes or modems<br>
<br>
I'm wondering if inserting an Asterisk server between PBX and PSTN would break
modems and faxes.<br>
For phones, I'm confident it will work.<br>
Any call that will cross Asterisk would be treated without any echo
cancellation or codec translation.<br>
<br>
So the main question is "is TE2XXP-enabled Asterisk able to switch TDM
without introducing latency that would spoil modem or fax services ?".<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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