[Asterisk-Users] Using Call Progress

[Kerry Garrison]

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.
 
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)
 
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.
 
Don't shoot the messenger.
 
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 -  <mailto:kerryg at techdatapros.com>
kerryg at techdatapros.com
 <http://www.techdatapros.com/> http://www.techdatapros.com 


  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
Buruschkin
Sent: Thursday, April 06, 2006 6:19 AM
To: Asterisk-Users
Subject: [Asterisk-Users] Using Call Progress


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.
 
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?
 
Any help would be appreciated!
 
Thanks!
 
- Eric Buruschkin
 

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