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<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2>Welcome to the painful world of analog phone lines. Unless
you are using a digital line, there really is no true call progress detection
available. In many situations this is not a problem, where we see this the most
is when you are trying to ring a zip device and a zap channel at the same time,
the zap call progress indicates an answered line the moment the zap channel goes
active, NOT when the far side answers. If you have a ring group with sip and zap
channels, what typically happens is that the sip phone will ring once, but as
soon as the TDM card places the outbound call, it is considered "answered" and
the sip phone stops ringing. Yes, you can enable callprogress and several other
tweaks but the end result is often the far side answering and Asterisk still
playing ring tones because there is no signal on the PSTN to indicate a far side
answer.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2>So, what to do when you find yourself in this situation and
adding a PRI is not a solution, the only way we have worked around this is to
make those outbound calls over a SIP or IAX service provider (and no, using a
SIP gateway like a Mediatrix 1204 does not solve the problem as it is a PSTN
issue)</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2>I know some people will argue this, but this was the result
of almost 12 hours of work with us and Digium to figure out this issue. After
MUCH debate and many hours of testing, this became the official
word.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2>Don't shoot the messenger.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=133065213-06042006><FONT face=Arial
size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Kerry
Garrison<BR>Director of Technical Services<BR></SPAN></FONT><STRONG><B><FONT
face=Arial size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Tech Data
Pros - <?xml:namespace prefix = st1 ns =
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w:st="on"><st1:PlaceName w:st="on">Orange</st1:PlaceName> <st1:PlaceType
w:st="on">County</st1:PlaceType></st1:place>'s Mobile IT Service
Provider<BR></SPAN></FONT></B></STRONG><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">(949) 502-7819 x200 - <A
title=mailto:kerryg@techdatapros.com href="mailto:kerryg@techdatapros.com"><EM
title=mailto:kerryg@techdatapros.com><I
title=mailto:kerryg@techdatapros.com><FONT title=mailto:kerryg@techdatapros.com
face=Arial><SPAN title=mailto:kerryg@techdatapros.com
style="FONT-FAMILY: Arial">kerryg@techdatapros.com</SPAN></FONT></I></EM></A><BR><A
title=http://www.techdatapros.com/ href="http://www.techdatapros.com/"><EM
title=http://www.techdatapros.com/><I title=http://www.techdatapros.com/><FONT
title=http://www.techdatapros.com/ face=Arial><SPAN
title=http://www.techdatapros.com/
style="FONT-FAMILY: Arial">http://www.techdatapros.com</SPAN></FONT></I></EM></A></SPAN></FONT>
</SPAN></DIV><BR>
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<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Eric
Buruschkin<BR><B>Sent:</B> Thursday, April 06, 2006 6:19 AM<BR><B>To:</B>
Asterisk-Users<BR><B>Subject:</B> [Asterisk-Users] Using Call
Progress<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>
<DIV><FONT face=Tahoma size=2>I'm attempting to use callprogress in my system,
and I'm having trouble. Callprogress always can tell if the line
is busy or ringing, but when the line is answered, the call does not get
bridged. Messages showing that "line is ringing" stop in the console and
if the called party hangs up, asterisk reports the line is busy.</FONT></DIV>
<DIV><FONT face=Tahoma size=2></FONT> </DIV>
<DIV><FONT face=Tahoma size=2>Are there any settings that I could use to help
with this issue? I am using asterisk 1.2.4 with TDM04B (FXO) cards on a
RHEL3 system. Something in indications.conf or zonedata.c/dsp.c in the
source that can be tweaked?</FONT></DIV>
<DIV><FONT face=Tahoma size=2></FONT> </DIV>
<DIV><FONT face=Tahoma size=2>Any help would be appreciated!</FONT></DIV>
<DIV><FONT face=Tahoma size=2></FONT> </DIV>
<DIV><FONT face=Tahoma size=2>Thanks!</FONT></DIV>
<DIV><FONT face=Tahoma size=2></FONT> </DIV>
<DIV><FONT face=Tahoma size=2>- Eric Buruschkin</FONT></DIV>
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