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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I've been playing with SIP a lot lately and we rolled
out some more of our employees on SIP based long distance (testing to see if it's
feasible to collocate our Asterisk at our data center with a PRI - hence the
talks before about sharing PRI, etc.)..</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I'm seeing an odd issue, I'm testing through IConnectHere
with limited success - however I seem to have one thing that may be reproducible
(too early to say if reproducible):</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> Layout:
POTS based phone, hooked to Zhone Channel bank - Zhone to Quad T1 Digium
Card on Asterisk server. Asterisk then Dial/SIP to analog based phone
system. </span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Trying: Analog phone system (called
party) says press 123 for sales.. We press 123. Suddenly call quality is
horrible (we can hear fine about 75% of the time, other 25% no audio, called
party says can't hear us hardly) - other DTMF tones don't
seem to go through right (maybe unclear so not being recognized).</span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Works: Calling what I would guess is
another digital PBX works fine using this same layout. I've tried a
few banks / large businesses and no problems.</span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Works: Can call a phone via SIP and
recognize DTMF from called party (i.e. # to transfer)</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Any ideas?</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><br>
We should have another SIP based gateway up next week and we now have pricing
on DC based PRI that's good so we may just go with PRI. I'd
rather avoid the cross connect fees / loops on the PRI since we can get SIP
based easier and have plenty of bandwidth - it seems like just another
thing to go wrong in Asterisk using PRI instead of Cisco delivered SIP (maybe I'm
backwards on this).</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>PS: How does IAX handle DTMF between Asterisk servers?</span></font></p>
<p><font size=2 face="Times New Roman"><span style='font-size:10.0pt'>Steve
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/ Virtual Dedicated / Dedicated Servers <br>
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size=2><span style='font-size:10.0pt'>, Inc. - </span></font><a
href="http://www.bitshop.com/"><font size=2><span style='font-size:10.0pt'>http://www.bitshop.com</span></font></a><font
size=2><span style='font-size:10.0pt'> - $149/month colo special</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
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