[Asterisk-Users] Dropped calls with analog lines using TDM400P

Wayne Wayne at planetWayne.com
Wed Nov 3 13:34:56 MST 2004


Hiya Andres,
I stared to have exactly the same problem - very soon after enabling the 
busydetect=yes in zapata.conf.
Used to work flawless with it set to 'no'. The only reason I turned it 
on was I was trying to busy line detection and auto redials. Ive set it 
back off at the mo - just to be sure that is the only problem.

TBH - Unless it drasticly alters the way * works for me (this is a very 
simple home set up here - nothing at all complex!!) I'll leave it set off...

That is unless someone else suggests what may be wrong...

Wayne.

Andres Maduro wrote:

>Hi, 
>
>I have successfully configured and built Asterisk and now it is working fine
>from the functionality point of view as sometimes we are getting dropped
>calls.
>
>The problem I am getting with POTS lines even if I receive/make a call from
>a sip or analog phone is that the call may be dropped randomly in the middle
>of the conversation, some times you can speak for 30 min, some times it
>drops or hang up at the start of the conversation.
>
>I have investigated in wiki and mailing lists deeply and have played with
>busydetect and callprogress in zapata.conf without success.
>
>I am currently using busydetect=yes with busycount=5 and I have tweaked the
>busy tone with Venezuela tone information and it works great.  I have
>deactivated callprogress according to wiki and comments on problem on non US
>lines.
>
>I have also found that modifying the rxgain and txgain seems to lower the
>number of false hangups but does not eliminate them.
>
>I read on the mailing list that there are some buggy TDM400P cards out there
>that cause random hangups.  How can I determine if this cards are the buggy
>ones or the problem is other ?
>
>Zttool reports Wildcard TDM400P REV E/F.
>
>Asterisk is running on Red Hat 9 dual Pentium III 400 Mhz with 128 Mbytes of
>RAM and 2 TDM400P cards with the following configuration:
>
>Asterisk version 1.0.1 with Zaptel driver version 1.0.0 
>
>-----------------------------------------------
>Zaptel.conf
>
>loadzone=us
>defaultzone=us
>
>fxols=1-4
>fxsls=5-6
>-----------------------------------------------
>Zapata.conf
>
>[channels]
>
>language = es1
>switchtype = national
>
>; EXTENSIONES INTERNAS
>
>context = internal
>group = 1
>signalling = fxo_ls
>
>echocancel = yes ; You can set this to 32, 64, or 128, tweak to your needs.
>echocancelwhenbridged = yes
>echotraining = 800 ; Asterisk trains to the beginning of the call, number is
>in milliseconds
>
>flash=750
>rxwink=300
>
>callprogress = no      ; Solo funciona bien con lineas USA
>callerid = asreceived
>usecallerid=yes
>hidecallerid=no
>callwaiting=no
>usecallingpres=yes
>callwaitingcallerid=no
>threewaycalling=yes
>; Support flash-hook call transfer (requires three way calling) transfer=yes
>; Support call forward variable cancallforward=yes ; Whether or not to
>support Call Return (*69) callreturn=yes callerid=asreceived
>
>
>;Sets whether asterisk should answer the channel immediately upon pickup,
>without waiting for input.
>; Esta debe estar en no.
>immediate=no
>
>callgroup=1
>pickupgroup=1
>
>; Extensiones internas analogicas iCONOS
>
>mailbox = 100
>callerid = Mercedes Lembert <100>
>channel => 1 
>
>mailbox = 102
>callerid = Ricardo Maduro <102>
>channel => 2
>
>mailbox = 103
>callerid = Fernando Guerrero <103>
>channel => 3
>
>mailbox = 104
>callerid = Laboratorio <104>
>channel => 4
>
>; I have used ztmonitor to tweak this values rxgain=11.0 txgain=-4.0
>
>;CANTV telco
>group=2
>signalling=fxs_ls
>
>faxdetect=both
>echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
>echocancelwhenbridged=yes
>echotraining=800 ; Asterisk trains to the beginning of the call, number is
>in milliseconds
>
>busydetect=yes
>busycount=5
>
>
>callprogress = no      ; Solo funciona bien con lineas USA
>callerid = asreceived
>usecallerid=yes
>hidecallerid=no
>callwaiting=no
>usecallingpres=yes
>callwaitingcallerid=no
>threewaycalling=yes
>; Support flash-hook call transfer (requires three way calling) transfer=yes
>; Support call forward variable cancallforward=yes ; Whether or not to
>support Call Return (*69) callreturn=yes callerid=asreceived
>
>context=pstn ; Points to the default context of your extensions.conf 
>
>channel => 5-6 ; Again X is the number of FXO modules you have
>-------------------------------------------------------------
>
>Indications.conf
>
>[general]
>country=ve
>
>[ve]
>description = Venezuela / South America
>ringcadance = 2000,4000
>dial = 426
>busy = 426/500,0/500
>ring = 426/1000,0/4000
>congestion = 480+620/250,0/250
>callwaiting = 440/300,0/10000
>dialrecall =
>!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
>record = 1400/500,0/15000
>info = !950/330,!1400/330,!1800/330,0
>
>---------------------------------------------------------------------
>
>Dmesg on Zaptel driver load
>Zapata Telephony Interface Registered on major 196 Freshmaker version: 71
>Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module
>1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3:
>Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4
>modules) Freshmaker version: 71 Freshmaker passed register test Module 0:
>Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode)
>Module 2: Not installed Module 3: Not installed Found a Wildcard TDM:
>Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States /
>North America)
>
>------------------------------------------------------------------------
>
>If you see any unusual thing or have any suggestion on what to try, please
>let me know.'
>
>If you need an additional conf file, please let me know so I can provided to
>you.
>
>Best regards and thanks in advance, 
>						Andres Maduro
>
>
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