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@default
context=default
[4002]
type=friend
userid=4002
secret=4002
host=dynamic
defaultip=192.168.0.202
mailbox=4002@default
context=default
[4003]
type=friend
secret=4003
host=dynamic
defaultip=192.168.0.203
mailbox=4003@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
__________________________________
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Do you need the busy detect feature as this often can cause random drop outs. Try changing busydetect=yes to busydetect=no in zapata.conf. Assuming that asterisk still works (i.e. it detects when people hang up and when lines are busy (most lines will do it ok but some won't (e.g. ntl cable in the UK from personal experience)) then that *should* solve your problem, if not then something more serious is going on. Shahid Mahmood wrote:> 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 >< SNIP > -- Alex Brett alex.brett@loho.co.uk +44 (0)870 744 2170 http://www.loho.co.uk/
Shahid, Looking below here are a few thoughts: Why are you defining switchtype and signalling for a switch? You don't need these for an X100P card. check your card make sure it's on it's own interrupt (/proc/interrupts) add echotraining=1 with your other echo commands. I had a problem like this previously that turned out to be an echo cancel problem. Still not sure where the problem was, but adding echotraining=1 fixed my problem. -Chris On 12:38 PM 4/23/2004, Shahid Mahmood wrote: >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 > <snip> >============= ../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 <snip> >; Support call forward variable >; >cancallforward=yes >callreturn=yes >echocancel=yes >echocancelwhenbridged=yes
Set callprogress=no and busycount=6 or higher in /etc/asterisk/zapata.conf On Fri, 2004-04-23 at 14:38, Shahid Mahmood wrote:> 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@default > context=default > > [4002] > type=friend > userid=4002 > secret=4002 > host=dynamic > defaultip=192.168.0.202 > mailbox=4002@default > context=default > > [4003] > type=friend > secret=4003 > host=dynamic > defaultip=192.168.0.203 > mailbox=4003@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@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-- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."
Alex, Chris and Eric: Based on your kind suggestions, I ade the following modifications in the zapata.conf: 1. busydetect=no 2. commented out busycount=xx 3. comented out switchtype=national I did make a 30 minute and an hour call on two servers during the day before modifications. Did not disconnect. I made the changes and now I am going to make a all night call (5,6 hours). Last but not the least, let me thank you al for your quick and right to the point responses. This was my first post (and first visit !) to this mailing list and I am extremely impressed by the response from the members. Earlier I used a commercial PBX, and it took me a week to get the response to even get the thing started ! I will keep you informed about the results. Thanks !! -shahid Eric Wieling <eric <at> fnords.org> writes:> > Set callprogress=no and busycount=6 or higher in > /etc/asterisk/zapata.conf > > On Fri, 2004-04-23 at 14:38, Shahid Mahmood wrote: > > 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. > > ....."Shahid Mahmood" <zshahid@yahoo.com> wrote in message news:20040423193857.89952.qmail@web90001.mail.scd.yahoo.com...> 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@default > context=default > > [4002] > type=friend > userid=4002 > secret=4002 > host=dynamic > defaultip=192.168.0.202 > mailbox=4002@default > context=default > > [4003] > type=friend > secret=4003 > host=dynamic > defaultip=192.168.0.203 > mailbox=4003@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@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 >