Re: [Asterisk-Users] zaphfc troubles

2005-05-18 Thread sjaak imap
Dear Nicolas Olivier
Just try the florz patch at http://zaphfc.florz.dyndns.org/
and look at cat /proc/interupts if your not sharing irq's
Maybe this will help
Good luck
Sjaak
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on zaptel, zaphfc, zaprtc:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo 
0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e
I'm using zaprtc as the gateway is running on a VIA motherboard without USB 
controller.
When I'm doing ztcfg -vv:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
Here are my confs:
/etc/zaptel.conf:
loadzone=fr
defaultzone=fr
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
/etc/asterisk/zapata.conf:
[channels]
language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2
/etc/asterisk/modules.conf:
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so
noload = chan_alsa.so
noload = chan_oss.so
[global]
chan_modem.so=yes
chan_zap.so=yes
The problem is that after ztcfg ran, I've got the following logs:
Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623, wanted 8 
got 7), probably a buffer overrun.
zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156, wanted 8 
got 7), probably a buffer overrun.
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
And when I start asterisk -c, same logs keep on, and I've finally a kernel 
crash:
Unable to handle kernel paging request at virtual address fffc
printing eip:
c0113cc0
*pde = d063
*pte = 
Oops: 
CPU:0
EIP:0010:[c0113cc0]Not tainted
EFLAGS: 00010013
eax: c248015c   ebx:    ecx: 0001   edx: 0001
esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
ds: 0018   es: 0018   ss: 0018
Process sshd (pid: 146, stackpage=c2c8f000)
Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248 c3819545
   0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4 0086
   c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008 c270c800
Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7] [c383cd78]
 [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd] [c0109f78]
 [c010c328]
Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing
Here is the output from asterisk:
No entry for terminal type screen;
using dumb terminal settings.
 == Parsing '/etc/asterisk/asterisk.conf': Found
 == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
 == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
 == Manager registered action Ping
 == Manager registered action Events
 == Manager registered action Logoff
 == Manager registered action Hangup
 == Manager registered action Status
 == Manager registered action Setvar
 == Manager registered action Getvar
 == Manager registered action Redirect
 == Manager registered action Originate
 == Manager registered action Command
 == Manager registered action ExtensionState
 == Manager registered action AbsoluteTimeout
 == Manager registered action MailboxStatus
 == Manager registered action MailboxCount
 == Manager registered action DBget
 == Manager registered action DBput
 == Manager registered action DBdel
 == Manager registered action ListCommands
 == Parsing '/etc/asterisk/manager.conf': Found
Asterisk Management interface 

Re: [Asterisk-Users] Recording in a call center

2005-04-29 Thread sjaak imap

You need something like this ??
exten = _0.,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial,SIP/[EMAIL PROTECTED]
and mount another server with NFS or SAMBA on /var/spool/asterisk/monitor
That would be the job.
Sjaak


I would like to record two months of calls.  The call center does not 
have a huge volume, probably like 60 calls a day and average about 15 
min a call.  I am using a quad port e1 card from digium.  i would like 
to record the calls on a seperate server than the one running asterisk 
to avoid any problems. 
 
any ideas?


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Re: [Asterisk-Users] .call MeetMe

2005-01-06 Thread sjaak imap
Arghh sure
Something like
Channel: ZAP/g1/xx/ZAP/g1/xxx/ZAP/g1/ etc 
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: monitor
Extension: 601 -- first meetme room
Priority: 1
ZAP/g1 is an ISDN30
But how do I know this 601 room is free
I've rooms 601 602 and 3,4,5,6,7,8,9 and so on
I like to pick up the first room who's empty.
Thanks
Florian Overkamp
Hi, 

 

Would it be possible to dail out to lett's say to 4 people 
with a .call 
file and put them directly into a free meetme room.
   

You could create an alias that dials 4 people at once, or you could generate
4 different call files. Seems to be easy enough..
Florian
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Re: [Asterisk-Users] Zaptel ISDN BRI settings for The Netherlands KPN

2004-12-29 Thread sjaak imap
Hello (Hallo) Remco Barende
Just look at http://www.ip-phone-forum.de/forum/portal.php it's in 
German language but for a dutch guy no problem I gues ?

Sjaak

Hi list!
I am installing an * box that will be installed on a site with KPN BRI 
ISDN in The Netherlands. I am using bristuff fron Junghanns.

Does anybody know the correct settings for this? I will not have 
internet access there which makes it harder to google around on location.

switchtype = euroisdn
is pretty obvious but what about these settings:
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net
pridialplan=local
prilocaldialplan=local
; trust user provided callerid (clip no screening)?
pritrustusercid = yes
immediate=yes
channel = 1-2
Thanks!!
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[Asterisk-Users] Install X-lite automatic with (windows) .ins file

2004-11-12 Thread sjaak imap
Hello
Maybe offtopic but in relation to asterisk.
I like to create just as i've done before for dialin internet 
connections settup.
Creating a ins file(windows) with php to give my user an easy setup with 
x-lite.
This ins file push the vars into the windows registry.

Maybe someone like to help me to crack down some problems.
I don't know the pasword format it's not  md5 but (the password in 
the example below is 123456789)
the RequestLicensKey and the Hash can I just leave as it is now ?.

I will set the php (click to install) script online, If more people like it
Thanks
Sjaak
[HKEY_CURRENT_USER\Software\XtenNetworksInc\X-Lite]
general:UpgradedRegistryAtPrivateBuildNumber=1103m
audio_tuning_wizard:ShownAudioTuningWizardOnce=dword:0001
settings:MicrophoneLevel=dword:0064
settings:SpeakerLevel=dword:0064
general:Username=50
general:AuthorizationUsername=50
general:ProxyServerEnabled=dword:0001
background:x=dword:01c3
background:y=dword:0035
general:RequestLicenseKeyID=1613CF1A5820402089FD7C07DC4E7AEF
general:RequestLicenseKeyHash=d2b1d2925b1a40bb4195765aa4ec728c67a42c62
menu_dialed_calls:Connected1=dword:
general:OutboundSIPProxy=
general:ShouldUseOutboundSIPProxy=dword:
iLBC:Enabled=dword:0001
g711a:Enabled=dword:
gsm:Enabled=dword:0001
settings:StartMinimized=dword:0001
exit_tip_dialog:user_said_dont_show_again=dword:0001
general:Domain=192.168.1.1
general:ProxyServer=192.168.1.1
general:DisplayName=Office
hosts:ListenSIPPort=dword:13c4
hosts:ListenSIPAddress=192.168.1.1
g711u:Enabled=dword:0001
general:Password=hex:a2,f0,f6,c1,f9,c4,a8,b9,12,35,ce,ad,72,44,24,f0,2b,29,\
 
2c,50,2b,b2,1e,2b,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,\
 
4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,\
 
3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,\
 
2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,\
 
95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,\
 
a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,\
 
2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,\
 
d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,\
 
23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,\
 
c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,\
 
d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,\
 
0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,\
 
31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,\
 
b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,\
 
f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,\
 
dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,\
 
38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,\
 
1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,\
 
bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,\
 
a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,\
 
93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,\
 
40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,\
 
e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,\
 
b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,\
 
f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,\
 
4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,\
 
3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,\
 
2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,\
 
95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,\
 
a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,\
 
2e,d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,\
 
d5,23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,\
 
23,c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,\
 
c4,d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,\
 
d6,0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,\
 
0e,31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,\
 
31,b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,\
 
b8,f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,\
 
f6,dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,\
 
dc,38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3,2e,d5,23,c4,d6,0e,31,b8,f6,dc,\
 38,1b,bc,a0,93,40,e0,b6,f6,4e,3f,2e,95,a3

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[Asterisk-Users] ISDN asterisk ISDN PBX possible ?

2004-10-20 Thread sjaak imap
Hello
May be this question is a little off topic.
I like to use asterisk as follow, is that possible.
NT1 ISDN  from telecom((isdn 1st card ) ASTERISK (2th isdn card)) 
  excisting regular PBX system.
In this way I don't have to invest in phone hardware stuff and i can 
join to the low cost VOIP calls.
Specialy for my  tele workers.

Incomming calls go transparantly through the asterisk server.
Outgoing calls go through  asterisk to internet VOIP provider if cheaper.
I'm mostly wurry about ISDN NT1 etc.
Maybe someone can point me to a allready excist lowcost device on the 
market.

Thanks
Sjaak
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