[Asterisk-Users] Re: PSTN Call drops randomly

Shahid
Fri, 23 Apr 2004 22:05:01 -0000

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
>
>
>
>
>
>
>
>
>
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