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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>I am fairly new at working with Asterisk.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>I am having a call quality issue that I really need to get
ironed out before we go to rollout the system in a week.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>Any help would be greatly appreciated!!! Even if it is just
pointing me in the right direction.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>My current setup:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>I have Asterisk Setup on a Dell Server. It has 2 T100P
cards. One will be for out T1 PRI from the Phone Company (We don’t have
this installed yet)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>The other T100P connects to a VINA T1 IAD (Channel Bank)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>I also have a Cisco 7960 SIP Phone attached and registered.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>The Server is connected to a broadband connection.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>My issue:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>When I call the IAX Demo from the SIP Phone, the call is
perfect. Asterisk Voice is 100%, and the Voice from the Digium Test server is
almost 100% (an occasional stutter)..but very usable.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>When I call the IAX Demo from a Phone connected to the VINA
Channel Bank, the Asterisk Voice is 100%, but once it connects to the test
server it is extremely choppy. You can kind of understand what is being said,
but it is very very poor quality and quite unusable.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>When I between the Channel Bank and the SIP phone, the
quality is 100% no problems at all.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>So..why does the VINA Channel Bank connection not seem to
like the IAX side of things, When I know that the IAX side is functioning great
when used from a SIP Phone?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>I don’t know what details would be pertinent to this,
but here is what the Asterisk Console Displays:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Executing
Playback("SIP/200-ad26", "demo-abouttotry") in new stack<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Playing 'demo-abouttotry' (language
'en')<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Executing
Dial("SIP/200-ad26",
"IAX2/guest@misery.digium.com/s@default") in new stack<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Called
guest@misery.digium.com/s@default<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Call accepted by 216.207.245.8 (format
gsm)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Format for call is gsm<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- IAX2/216.207.245.8:4569-1 is ringing<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- IAX2/216.207.245.8:4569-1 answered
SIP/200-ad26<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Hungup 'IAX2/216.207.245.8:4569-1'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> == Spawn extension (from-sip, 861, 2) exited non-zero
on 'SIP/200-ad26'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>asterisk1*CLI><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>asterisk1*CLI><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Starting simple switch on 'Zap/25-1'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Executing
Playback("Zap/25-1", "demo-abouttotry") in new stack<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Playing 'demo-abouttotry' (language
'en')<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Executing Dial("Zap/25-1",
"IAX2/guest@misery.digium.com/s@default") in new stack<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Called
guest@misery.digium.com/s@default<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Call accepted by 216.207.245.8 (format
gsm)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Format for call is gsm<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- IAX2/216.207.245.8:4569-2 is ringing<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- IAX2/216.207.245.8:4569-2 answered
Zap/25-1
<---------- THIS IS WHERE THE AUDIO BECOMES ALL CHOPPED UP.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Hungup 'IAX2/216.207.245.8:4569-2'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> == Spawn extension (chan_bank, 861, 2) exited
non-zero on 'Zap/25-1'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'> -- Hungup 'Zap/25-1'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:11.0pt;
font-family:Arial'>asterisk1*CLI><o:p></o:p></span></font></p>
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