[Asterisk-Users] PSTN Call drops randomly - Email found in subject
Greg Scasny
gscasny at golden-tech.com
Fri Apr 23 13:43:02 MST 2004
Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-----Original Message-----
From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Shahid Mahmood
Sent: Friday, April 23, 2004 2:39 PM
To: asterisk-users at lists.digium.com
Subject: [SPAM] - [Asterisk-Users] PSTN Call drops randomly - Email found in subject
Dear List members,
After succesfully installing the * on a couple of systems, and putting
them on test, I observed that there is an intermittent call drop on
PSTN line.
The systems are
- Dell Optiplex P3/500MHz/128MB
- Built-in ethernet
- 1 X100P (Motorolla chip) card on PCI
- 10G HDD etc.
- Asterisk April 17 CVS.
- 2 Mediatrix FXS ATA (4 phones)
- 2 Grandstream phones.
- sip.conf, zaptel.comnf and zapata.conf included below
Also let me know what do I need to "turn on" to get fine details about
the event when it happens.
Any help will be greatly appreciated.
Regards.
-shahid
========== sip.conf=============
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
;srvlookup = yes ; Enable DNS SRV lookups on outbound
; Asterisk only uses the first host in
;pedantic = yes ; Enable slow, pedantic checking for
tos=lowdelay ; IP QoS parameter, either keyword or
; like tos=184
;maxexpirey=3600 ; Max length of incoming registration
we allow
;defaultexpirey=120 ; Default length of incoming/outoing
registratio
n
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;videosupport=yes ; Turn on support for SIP video
externip = xxxxxxxxxxxxxxxxx ; Address that we're going to put in
; if we're behind a NAT
localnet = 192.168.0.0 ; Internal NETWORK address
localmask = 255.255.255.0 ; Internal netmask
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[4001]
type=friend
secret=4001
host=dynamic
defaultip=192.168.0.201
mailbox=4001 at default
context=default
[4002]
type=friend
userid=4002
secret=4002
host=dynamic
defaultip=192.168.0.202
mailbox=4002 at default
context=default
[4003]
type=friend
secret=4003
host=dynamic
defaultip=192.168.0.203
mailbox=4003 at default
============= ../zaptel.conf (uncommented lines) =============
fxsks=1
loadzone = us
defaultzone=us
============= zaptel.conf =======================
;
; Zapata telephony interface
;
; Configuration file
[channels]
;
; Default language
;
;language=en
;
; Default context
;
context=default
;
switchtype=national
signalling=fxo_ls
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
;
; Support call forward variable
;
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1-2
immediate=no
busydetect=yes
;
busycount=4
musiconhold=default
;
jitterbuffers=8
context=bell
signalling=fxs_ks
callerid=asreceived
channel=1
; --- uncomment for second card
;signalling=fxs_ks
;callerid=asreceived
;channel=2
__________________________________
Do you Yahoo!?
Yahoo! Photos: High-quality 4x6 digital prints for 25¢
http://photos.yahoo.com/ph/print_splash
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at 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
More information about the asterisk-users
mailing list