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Andres Maduro

2004-Nov-03 13:09 UTC

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

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

Brancaleoni Matteo

2004-Nov-03 13:20 UTC

head link

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

hi

Il mer, 2004-11-03 alle 21:09, Andres Maduro ha scritto:> I am currently using busydetect=yes with busycount=5 and I have tweaked the
raise busycount to at least 6 or even 8.

In my experience 4 is an assurance to see dropped calls,
5 also, but less frequently, 6 never.
but I prefer 8, to be 100% sure.

matteo.

-- 
Brancaleoni Matteo <mbrancaleoni@espia.it>
Espia Srl

Wayne

2004-Nov-03 13:34 UTC

head link

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

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
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

asterisk users - Nov 2004 - Dropped calls with analog lines using TDM400P

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

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

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