[Asterisk-Users] dial (iax/Xsip/y) get y fraction earlier

2005-09-23 Thread Sjaak Nabuurs

Hello


I like to call to 2 providers
provider X = IAX
provider Y = SIP


exten = _06.,1,Dial(IAX2/X/${EXTEN},30,r)(SIP/[EMAIL PROTECTED]) 
exten = _06.,2,Hangup 


Provider X is working but provider Y never shows up.
What's wrong ??

How can I get provider Y working a fraction earlier the provider X



Thanks


Sjaak




___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Asterisk RSS list feeder ready

2005-06-28 Thread Sjaak Nabuurs

Hello


Just for fun a rss newsreader for the asterisk users and biz list.
Easy to use and now with the complete history to search.

Just use it if you like

Thanks


Sjaak

___
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] Asterisk rss list feeder ready

2005-06-28 Thread Sjaak Nabuurs

Sorry URL is needed


http://asterisk.voipexco.com


___
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] AreskiCC + MySQL

2005-05-10 Thread Sjaak Nabuurs
Hello * Users
Did somebody get managed to get AreskiCC work under mysql.
If so is there anywhere to find the database structure for mysql.
Thanks
Sjaak
___
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] RSS feed Asterisk-Users

2005-04-28 Thread Sjaak Nabuurs
hello Asterisk-Users

I created just for fun a rss feed for Asterisk-Users and Asterisk-Biz list
First for myself but if it is usefull for you can use it if you like.

But some questions
Is it allowed ?
If you need add-on's please let me know.
When many people will use it I need to generate a little money to pay
traffic is it allowed to ad googleadd's ?

I've added a search box and later on I will add the whole list history so
it will be usefull to search.

Just look at http://asterisk.voipexco.com


Thanks, and have fun


Sjaak

___
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] bad sound ISDN bristuff

2005-02-12 Thread Sjaak Nabuurs
Hello * users
I've problems with sound quality on zaphfc
Asterisk works fine good sound quality.
If I do make load in the bristuf.xx zaphfc dir then sound quality 
drops directly.
Even if I don't load the chan_zap  in the modules.conf

I use this config on more (even old 400Mhz machines) and works correctly.
Looks like an hardware problem but I can't find it.
I don't see any conflics on IRQ or interupts
Using :
Asterisk 1.0.1-BRIstuffed-0.2.0-RC1
ISDN HFC cologne cards
If you have any solutions I would like to hear it.
cat /proc/interupts
---
   CPU0
  0:   32065843  XT-PIC  timer
  1:  2  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  6:  0  XT-PIC  ehci_hcd
  8:  1  XT-PIC  rtc
 10:  757932899  XT-PIC  zaphfc
 12:   14254461  XT-PIC  eth0
 14:1259128  XT-PIC  ide0
NMI:  0
ERR:  0
cat /proc/pci
-
PCI devices found:
  Bus  0, device   0, function  0:
Host bridge: Silicon Integrated Systems [SiS] SiS651 Host (rev 2).
  Master Capable.  Latency=32.
  Non-prefetchable 32 bit memory at 0xe800 [0xebff].
  Bus  0, device   1, function  0:
PCI bridge: Silicon Integrated Systems [SiS] SiS 530 Virtual 
PCI-to-PCI bridge (AGP) (rev 0).
  Master Capable.  No bursts.  Min Gnt=8.
  Bus  0, device   2, function  0:
ISA bridge: Silicon Integrated Systems [SiS] SiS962 [MuTIOL Media 
IO] (rev 37).
  Bus  0, device   2, function  5:
IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev 0).
  IRQ 11.
  Master Capable.  Latency=128.
  I/O at 0xa400 [0xa40f].
  Bus  0, device   3, function  0:
USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller 
(rev 15).
  IRQ 3.
  Master Capable.  Latency=32.  Max Lat=80.
  Non-prefetchable 32 bit memory at 0xe700 [0xe7000fff].
  Bus  0, device   3, function  1:
USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller 
(#2) (rev 15).
  IRQ 5.
  Master Capable.  Latency=32.  Max Lat=80.
  Non-prefetchable 32 bit memory at 0xe680 [0xe6800fff].
  Bus  0, device   3, function  3:
USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller 
(rev 0).
  IRQ 6.
  Master Capable.  Latency=32.  Max Lat=80.
  Non-prefetchable 32 bit memory at 0xe600 [0xe6000fff].
  Bus  0, device   4, function  0:
Ethernet controller: Silicon Integrated Systems [SiS] SiS900 10/100 
Ethernet (rev 145).
  IRQ 12.
  Master Capable.  Latency=32.  Min Gnt=52.Max Lat=11.
  I/O at 0x8800 [0x88ff].
  Non-prefetchable 32 bit memory at 0xe580 [0xe5800fff].
  Bus  0, device  14, function  0:
Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 2).
  IRQ 10.
  Master Capable.  Latency=16.  Max Lat=16.
  I/O at 0x8400 [0x8407].
  Non-prefetchable 32 bit memory at 0xe500 [0xe5ff].
  Bus  1, device   0, function  0:
VGA compatible controller: Silicon Integrated Systems [SiS] 
SiS65x/M650/740 PCI/AGP VGA Display Adapter (rev 0).
  IRQ 11.
  Prefetchable 32 bit memory at 0xf000 [0xf7ff].
  Non-prefetchable 32 bit memory at 0xe780 [0xe781].
  I/O at 0xd800 [0xd87f].


Thanks anyway
Sjaak
--
Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden.
___
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] FAX detection in extentions.conf

2005-01-19 Thread Sjaak Nabuurs
Hello
I have a PBX for our company with in front a ISDN * ISDN server.
I would like to detect if the outgoing  line is a fax call or  voice call.
I know ther's a fax detection in zaptel.
Now the question can I make desision in extentions.conf when * detects a 
fax call.

FAX  PBX  ZAP/G1  ZAP/G2the world
and not
FAX  PBX  ZAP/G1  IAX2   voip world
Thanks
___
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] .call MeetMe

2005-01-06 Thread Sjaak Nabuurs
Hello
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.

Thanks
Sjaak
___
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] Help me ($$$) with install h323

2004-12-20 Thread Sjaak Nabuurs
Hello
Does anybody who have experience in installing the h323 modules in asterisk.
I try'd it many times an spend xxx hours to  install it but didn't get 
lucky so far.

I have asterisk 1.0.1 running with bristuffed  2.0
I'm willing to pay for this.
Sjaak


___
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] ISDN on com port /dev/ttyS0 possible ??

2004-12-07 Thread Sjaak Nabuurs
Hello
I buyed a new server 2*XEON in a 2inch High 19case.
Now I have a problem that the riser card is 64bit so an ISDN PCI modem isn't 
possible.
My question is can I use ISDN on com port /dev/ttyS0.
If yes can I use it like the example in modem.conf as /dev/ttyI0 but use 
/dev/ttyS0
Does anybody have expirience with this ?
Thanks
Sjaak

--
Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] CLI h.323 show codecs shows nothing

2004-11-23 Thread Sjaak Nabuurs
Hello
I like to make calls to an h.323 device.
I'm using Nuphone h323.
Compiled everything okay I Guess
When I make a connection * SIP  h323 device, the phone is ringing and then * tells me 
No one available. and disconnect
Thinking this is a codec problem and check in CLI h.323 show codecs and * shows 
nothing.
I try many combination in the h323.conf like.
allow=all
or
disallow=all
allow=ulaw
allow=gsm
Could that be the problem to connecting to an h.323 device ?
Can somebody give me a good h323.conf example for outgoing calls to an h323.
Or any advice would be nice.
I'm getting @#!$$%^#^$%^$$%
Cheers
Sjaak

--
Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] if NOT SipUser then Dial(Zap/1/${EXTEN})

2004-11-16 Thread Sjaak Nabuurs
Hello
I like to setup a dial plan where sip users have the same number as 
there normal phone number.
So if sombody dial the number i like that asterisk looks first is it a 
number who exsist in MySQL sipfriends if not dial out with a ZAP device.

I know that extention.conf doesn't have a programming language but 
something like this

if (   SIP/${EXTEN})
{
exten = _XX,1,Dial(SIP/${EXTEN})
exten = _XX,2,Hangup
}
else
{
exten = _XX,1,Dial(Zap/1/${EXTEN})
exten = _XX,2,Hangup
}
Does anybody here a solution for this.
Thanks.
Sjaak
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] Can some bady help me ???

2004-11-11 Thread sjaak nabuurs
Hello
Are you online with you suse box.
If you give me for a ssh login I can look to it.
Or otherwise give us more info.
Sjaak
Rodney Acosta Coya wrote:
I get the pakage asterisk-1.0.0.tar.gz but i cant compile a install it
For help, i have a suse 9.1 distro in my computer.
im new using linux
thanks in advance 
Rodney

 


--
Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] grandstream 102 flashing

2004-10-21 Thread Sjaak Nabuurs
Check out your DNS settings
Wait +60 sec
dean collins wrote:
Can you (or anyone else out there) tell me how to fix this? Basically 
what happened was I tried to log into the web interface (ip address 
allocated by sbs dhcp using mac address) but when I hit login it 
rebooted and has been doa since then).

I checked using MS sbs network monitoring all it seems to be doing is 
asking for a ARP Rarp request to 67.153.142.69

The other thing is it thinks it is ip address 192.168.1.160 but that 
isnt even part of my network.

Any thoughts on what to do from here? The lcd display is totally non 
responsive.

* From: * [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of 
*BetaTeilchen
*Sent:* Wednesday, October 20, 2004 11:06 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] grandstream 102 flashing

This flashing is an indicator for a damaged firmware in your phone. 
Maybe an interrupted TFTP-Download when powered up or just a wrong 
firmware.

dean collins schrieb:
Does anyone know what it means when a grandstream flashes the red key 
light 5 times repeatedly in cycles? I got a new handset delivered to 
me today, powered up fine until I tried to access it via the web 
interface using the password admin and then it rebooted with the lcd 
never displaying again and the red keys flashing 5 times then a break 
of 3 seconds then repeat.

Cheers,
Dean







___   

Asterisk-Users mailing list   

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]   

http://lists.digium.com/mailman/listinfo/asterisk-users   

To UNSUBSCRIBE or update options visit:   

   http://lists.digium.com/mailman/listinfo/asterisk-users   


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] codec problems with astcc and not with sip trhough aix

2004-10-20 Thread sjaak nabuurs
Hello .
I have a 800 tollfree trhough iax to my * server.
If I phone to 800 number to the * machine to a  sip phone everything is 
okay.
exten = 8,1,Dial(SIP/12345678,20)
   -- Accepting AUTHENTICATED call from xx.xx.xx.xx, requested format = 
4, actual format = 4

If I change my extention to the astcc script i get lott's of these errors
exten = 8,1,DeadAGI(astcc.agi)
Oct 21 00:58:22 NOTICE[1110648512]: channel.c:1314 ast_read: Dropping 
incompatible voice frame on IAX2/[EMAIL PROTECTED]:4569/1 of format 
G729A since our native format has changed to ULAW

I see these two difference are codec problems but why.
and what to do with this problems
When I phone with a second SIP phone to the 8XX number everything is 
okay to the sipphone and to the astcc script
My sip phones accept all codecs

My aix.conf looks like this
[tollfreeprovider]
type=user
host=xx.xx.xx.xx
secret=mysecret
notransfer=yes
disallow=all
;allow=g729
;allow=gsm
allow=ulaw
trunk=yes
context=tollfree
I don't know, could this be a problem for my tollfree number provider or 
is it me.

Cheers

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] ASTCC newbie

2004-10-20 Thread sjaak nabuurs
I just installed it 2 days ago and I was looking for a long time to get 
thes problem solved.

Be aware off creating the Mysql database the astcc.cgi script will do.
But I disable the create lines in the perl script.
The astcc.cgi script will endles wait.
Good luck
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [Asterisk-Users] SIP Extensions

2004-10-20 Thread sjaak nabuurs
Hello Ron Ramos
With the power of bash it's easy
copy this code to a file name it createsip.sh

#!/bin/bash
for ((i=2000 ; i  8000; i++ )); do
echo [$i]
echo secret=$i
echo type=friend
echo username=$i
echo 
done

chmod 700 createsip.sh
./createsip.sh  /etc/asterisk/sip.conf
and your done
Thats all

Hi All,
How can I be able to define multiple SIP extensions?
Do I have to define each extensions on sip.conf?
For example, extension 2000-8000, do I have to define it one by one
on sip.conf?
[2000]
secret=2000
type=friend
username=user2000
..
..
[2001]
secret=2001
type=friend
username=user2001
..
..
..
..
..
[8000]
Is there a way I can define multiple extensions
like...
[_2XXX]
..
..
..
[_8XXX]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] sip show peers MySQL Database

2004-10-13 Thread Sjaak Nabuurs
Hello
How can i see the sip show peers if I use sipfriends database.
I see only the peers who are in the sip.conf.
Thanks
Sjaak
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] SIP providers USA

2004-08-18 Thread Sjaak Nabuurs
Hello


This Question maybe ask many times.
I'm From the Netherlands and looking for an sip/aix  pstn provider in the
US.I asked wipphone, vonage but they don't work for europe.
I've visit many not finished website's about sip  pstn

Can somebody recomed me any good sip/aix provider in the us.


Thanks




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] The best simple solution (newbee)

2003-03-31 Thread sjaak nabuurs
Hi

I'm very new to astrisk and I like to ask you as users with expirience what
hardware to use and how to set this up.

What I like to create is the followe :

Use a normale phone dail a number  astrisk (2Mbit) Internet(64Kb) 
astrisk  Normal phone starts to ring and someone pick up the phone and
talk and also phone reverse.

Ther's so lot of hardware x100p phonejack linejack Cisco and more and more
i've not a good idea what's good for me.
I've read so much that now my brain is [EMAIL PROTECTED]*()_
My main language isn't English so I understand everything for 90% and that's
not enough to buy something like this.

I just like to setup a simple and (low budget) configuration.
I know that 64K line isn't that much but is it acceptable ?
If it's equeal to msn or messenger it's okay for me.
My telephone bill is rissing like the sun.
My familie oversea doesn't have any clue about computers so i like to setup
this in linux so I can manage it..

Thanks you very much for your advice.

Sjaak








--
Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users