date:20050911

Hi,

I've installed an TE406p, asterisk1.2 on tyan opteron
board.
After installation there is no interrupts from TE406p.
Is this board stable? 
Should i change * version to 1.0.9?

Regards,
Jason


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RE: [Asterisk-Users] TE406p no interrupts

2005-09-11 Thread Alexander Lopez

Did it take an interrupt??

Whats does /proc/interrupts say??

Did you check your span= settings in zaptel.conf??

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 5:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TE406p no interrupts

Hi,

I've installed an TE406p, asterisk1.2 on tyan opteron
board.
After installation there is no interrupts from TE406p.
Is this board stable? 
Should i change * version to 1.0.9?

Regards,
Jason


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Re: [Asterisk-Users] Integrating with existing analog PBX

2005-09-11 Thread John Daragon


Martin Allen wrote:

Asking about inserting Asterisk between a 4 line analogue PBX and the 
outside world...


The proposed solution (with 4 x FXO and 4 x FXS using 2 TDM400 cards) 
will work fine until the asterisk box dies or suffers power failure.


An alternative may be to use 4 Sipura SPA-3000 ATAs (which have an FXO 
and an FXS port as well as an RJ45 network port (think of them as two 
ATAs an a single box...) and are cheap (see http://www.voiptalk.org )



  PSTN

***||
* *  SIP to FXO  +---+
* Asterisk* -|   |
* *  |SPA-3000   |
* * -|   |
* *  SIP to FXS  +---+
***||

  PABX


In the event of power failure the FXO port is switched directly to the 
FXS port, effectively bypassing the IP side of things completely.


Actually, I think you *could* build what you're describing just with the 
SPA-3000s, but you would, of course, lose a lot of flexibility...



jd

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RE: [Asterisk-Users] TE406p no interrupts

What kernel are you using?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 7:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TE406p no interrupts

Hi,

I've installed an TE406p, asterisk1.2 on tyan opteron
board.
After installation there is no interrupts from TE406p.
Is this board stable? 
Should i change * version to 1.0.9?

Regards,
Jason


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RE: [Asterisk-Users] TE406p no interrupts


zaptel.conf

span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109

cat /proc/interrupts 
--
   CPU0   CPU1   
  0: 200570 273687IO-APIC-edge  timer
  4:  0 16IO-APIC-edge  serial
  8:  0  0IO-APIC-edge  rtc
  9:  0  0   IO-APIC-level  acpi
169:   1425   3343   IO-APIC-level  libata,
ehci_hcd, ohci_hcd, ohci_hcd
177:  0  2   IO-APIC-level  AMD
AMD8111, uhci_hcd, ohci1394
185:   1979 30   IO-APIC-level  uhci_hcd,
eth0
193:  4  4   IO-APIC-level  wct4xxp
NMI: 18 22 
LOC: 474031 474031 
ERR:  0
MIS:  0

Thanks.

--- Alexander Lopez [EMAIL PROTECTED] wrote:

 Did it take an interrupt??
 
 Whats does /proc/interrupts say??
 
 Did you check your span= settings in zaptel.conf??
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Jason Kim
 Sent: Sunday, September 11, 2005 5:48 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] TE406p no interrupts
 
 Hi,
 
 I've installed an TE406p, asterisk1.2 on tyan
 opteron
 board.
 After installation there is no interrupts from
 TE406p.
 Is this board stable? 
 Should i change * version to 1.0.9?
 
 Regards,
 Jason
 
 
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[Asterisk-Users] OpenH323-Channel Q.931-Problems with Gatekeeper

2005-09-11 Thread Sebastian Mangelkramer

Title: OpenH323-Channel Q.931-Problems with Gatekeeper






Dear Mailinglist-User

currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities.
SIP and ISDN works fine, but H.323 not.


In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the chan_oh323 (version 0.6.5).
We successfully tested in/egress calls without any problems.

But when we started to connect our Asterisk with the Gatekeeper (Siemens Surpass) of an big german Carrier
we noticed some strange problems we couldn`t solve until right now.

The registration with the gatekeeper is successful. But every from and to our PBX will be cleared/rejected by an Q.931 cause.


Our system-layout looks like:

Debian GNU/Linux 3.1 aka sarge with Kernel 2.6.12, i386
Asterisk 1.0.9 (stable)
Pwlib 1.16
OpenH323 1.13.5
Chan_oh323 0.6.6


Perhaps you know some problems with Asterisk and the H.323-Channel.
We tried to compile and test nearly every version of openh323 and chan_oh323, but it wasn`t successful.




Best regards from Germany,

Sebastian.



Nearby we will post our configs and logs:


1.) chan_oh323.conf
-

[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
language=de
; erweitertes logging aktivieren (debugging)
wrapLibTraceLevel=9
libTraceLevel=9
libTraceFile=/var/log/asterisk/oh323.log
; gatekeeper des carrier
gatekeeper=XXX.XXX.XXX.XXX
gatekeeperTTL=600
userInputMode=TONE
; detailierte cdr erstellen
amaFlags=billing
accountCode=0123456789
; eingehende calls an diesen context senden
context=carrier-in
[register]
context=carrier-in
alias=0123456789
[codecs]
codec=G711A
frames=20




2.) Status of OpenH323 channel driver
---


*CLI oh323 show conf

Version: 0.6.6
Listening on address: 0.0.0.0:1720
Gatekeeper used: [EMAIL PROTECTED] (Registered)
FastStart/H245Tunnelling/H245inSetup: ON/OFF/OFF
Supported formats in pref. order: alaw0
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: 2
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 10
Max call rate (ingress direction): 1.00/30
Default language:
Default music class:
Default context: h323-in



3.) Verbose debugging of OpenH323 channel driver while calling from carrier
---

*CLI [4]WrapH323EndPoint::CreateConnection: Creating a H323Connection [1797]
[4]WrapH323Connection::WrapH323Connection: WrapH323Connection created.
[2]WrapH323Connection::OnReceivedSignalSetup: Received SETUP message...
[2]WrapH323Connection::OnAnswerCall: User - (016097XX) [IP of Carrier-GK] is calling us...
[3]WrapH323Connection::OnAnswerCall: Call ID: 02cb6411-b5a7-178c-2499-0800062a0cf1
[3]WrapH323Connection::OnAnswerCall: Conference ID: 02cb6411-b5a7-178c-2499-0800062a0cf1
[3]WrapH323Connection::OnAnswerCall: Call reference: 1797
[3]WrapH323Connection::OnAnswerCall: Call token: ip$IP of Carrier-GK:36031/1797
[3]WrapH323Connection::OnAnswerCall: Call source alias: - (016097XXX) [IP of Carrier-GK](35)
[3]WrapH323Connection::OnAnswerCall: Call dest alias: 0123456789 0123456789 E164:123456789 ip$10.0.0.20:1720(64)
[3]WrapH323Connection::OnAnswerCall: Call source e164: 016097XX(12)
[3]WrapH323Connection::OnAnswerCall: Call dest e164: 0123456789(13)
[3]WrapH323Connection::OnAnswerCall: Call RDNIS: (0)
[3]WrapH323Connection::OnAnswerCall: Remote Party number: 016097
[3]WrapH323Connection::OnAnswerCall: Remote Party name: - (016097) [IP of Carrier-GK]
[3]WrapH323Connection::OnAnswerCall: Remote Party address: [EMAIL PROTECTED] of Carrier-GK:36031
[3]WrapH323Connection::OnAnswerCall: Remote Application: Surpass Siemens 4/130(21)
Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797' detected.
 -- Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797' detected.
Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797', channel 'OH323/[EMAIL PROTECTED] of Carrier-GK-d66d'.
 -- Inbound H.323 call 'ip$IP of Carrier-GK:36031/1797', channel 'OH323/[EMAIL PROTECTED] of Carrier-GK-d66d'.
[3]WrapH323EndPoint::OpenAudioChannel: Direction = RECODER, Buffer = 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-ALaw-64k
Setting channel 'OH323/[EMAIL PROTECTED] of Carrier-GK-d66d'

RE: [Asterisk-Users] TE406p no interrupts

I'm using FC3.

uname -a
-
Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2
15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux


zaptel.conf

span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109

cat /proc/interrupts 
--
   CPU0   CPU1   
  0: 200570 273687IO-APIC-edge  timer
  4:  0 16IO-APIC-edge  serial
  8:  0  0IO-APIC-edge  rtc
  9:  0  0   IO-APIC-level  acpi
169:   1425   3343   IO-APIC-level  libata,
ehci_hcd, ohci_hcd, ohci_hcd
177:  0  2   IO-APIC-level  AMD
AMD8111, uhci_hcd, ohci1394
185:   1979 30   IO-APIC-level  uhci_hcd,
eth0
193:  4  4   IO-APIC-level  wct4xxp
NMI: 18 22 
LOC: 474031 474031 
ERR:  0
MIS:  0

Thanks.

--- Boris Bakchiev [EMAIL PROTECTED] wrote:

 What kernel are you using?
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Jason Kim
 Sent: Sunday, September 11, 2005 7:48 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] TE406p no interrupts
 
 Hi,
 
 I've installed an TE406p, asterisk1.2 on tyan
 opteron
 board.
 After installation there is no interrupts from
 TE406p.
 Is this board stable? 
 Should i change * version to 1.0.9?
 
 Regards,
 Jason
 
 
 __
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RE: [Asterisk-Users] TE406p no interrupts

Well.
Try this please (but only if you're running on the latest sources).
Open wct4xxp.c sources and search for pci_module_init
Replace it with pci_register_driver
So the line should read:
res = pci_register_driver(t4_driver);

That allows you to get the card working on 2.6.13 in almost exactly the
same setup as yours.

One weird thing though. Do no use insmod ./wct4xxp.ko from zaptel
directory as it will not work. Do a proper make install and then
modprobe.


This is just part of the fixes you might need to do.
If you encounter a problem after span reconfiguration (ztcfg) let me
know.

If you get stuck.. let me know.

Regards

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 8:14 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] TE406p no interrupts

I'm using FC3.

uname -a
-
Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2
15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux
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[Asterisk-Users] Make asterisk call out

2005-09-11 Thread Andreas Moroder


Hello,

in our public hospital incase of emergency in the night or weekend we
must call many people.
Is it possible to use asterisk to call automatically a list of number.
The numbers should be called in a round-robin way as long as they don't
take up the phone and confirm by digitin in a code.

The calls should be started by an internal call to a certain number on
the asterisk server. It should be possible to have different lists of
numbers for different alarm levels.

Bye
Andreas


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Re: [Asterisk-Users] Ignore incomingcall?

2005-09-11 Thread David Cook

Use a separate context for each Dring.

dring2 cadence 0,0,0 will identify the primary number not the secondary.
 If you want dring1=main number  dring2=distinctive ring num then you
need dring1 as 0,0,0 and dring2 as the alternate cadence.

This context will ignore the calls on the main number if dring1context
is set to primary in zapata.conf.

[primary]
exten = s,1,NoOp(${CALLERID})
exten = s,2,Hangup



 Is there a way to tell asterisk to ignore an incoming call?
 I am using distinctinveringdetection and I am only interested in
 answering
 calls
 on the 2nd number.  Any call to the main line should just be ignored.

 right now I have a context set for dring2 cadence 0,0,0
 exten = s, 1, wait(30
 exten = s, 2, Hangup

--
David Cook
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Re: [Asterisk-Users] Make asterisk call out

2005-09-11 Thread Brian Roy


On 9/11/05, Andreas Moroder [EMAIL PROTECTED] wrote:
Is it possible to use asterisk to call automatically a list of number.


Yes, it's possible. It will require a little effort to do some scripting, but not much. They key to making Asterisk call out, will be using the call files.

Here is the wiki page http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

With a little effort you can do this yourself, or request someone write something for you on the -biz list. Someone would do it for a few bucks.

-Brian


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Re: [Asterisk-Users] VoipBuster again -- WORKS NOW!!!!

2005-09-11 Thread Rudolf Ladyzhenskii


Hi, all

Finally got it to work.

TWo problems.
1. Stupid erro on firewall caused authentication failure. BAsically IAX 
ports were forwarded to the * box without noting the interface they were 
coming on. Thsi is OK when external clients tried to connect, but when I 
tried to connect to outside, firewall was forwarding my requests back to 
asterisk box, so it was trying to authenticate against itself. Runnign 
Ethereal on firewall helped to find this problem.


2. Still could not connect call, although registration was working.
Had to change dialing string to:
exten = _0.,2,Dial,IAX2NAME:[EMAIL PROTECTED]/00613${EXTEN:1}

By some reason I had to use full server name, it was not picking it up from 
iax.conf Will look at it later when I have a chance.


Rudolf


- Original Message - 
From: Sander [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Sunday, September 11, 2005 5:26 PM
Subject: RE: [Asterisk-Users] VoipBuster again



Try this ip for register something looks wrong with iax.voipbuster.com
I changed it a while ago because i had some dns problems in with my 
provider
and this ip came up when i pinged now you can't ping to the adress and 
it's

another ip


register = username:[EMAIL PROTECTED]




-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf 
Ladyzhenskii

Verzonden: zondag 11 september 2005 0:25
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] VoipBuster again

Here is what I get when reloading IAX2:

Not every time, though
 == Parsing '/etc/asterisk/iax.conf': Found Sep 11 08:48:29 WARNING[3240]:
chan_iax2.c:5402 iax2_register: Host 'iax.voipbuster.com' not found at 
line

164

Strange, because name resolves to IP address.

Ok, I reload IAX2 again and no more warning.
Then it tries to register and fails:

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: 
LAGRQ

  Timestamp: 10018ms  SCall: 1  DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: 
LAGRQ

  Timestamp: 10018ms  SCall: 1  DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
  Timestamp: 00017ms  SCall: 1  DCall: 0 [213.61.187.146:4569]
  USERNAME: USERNAME
  REFRESH : 60

And so it goes.

Call then fails too...

I am suspecting two things:
1. I am starting to wonder if registering a user in Australia using
VoipBuster application does not create an IAX account
Can someone who has an IAX account try creating one for me? Bogus name and
password. my e-mail is [EMAIL PROTECTED]

2. Firewall ports are not open. I am sure all the right ports are 
forwarded

to my * box (5060, 4569, 1-2).
I will set up ethereal on my firewallbox to see what comes out to the www
and what comes back.


Thanks,
Rudolf

- Original Message -
From: Sander [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, September 10, 2005 11:32 PM
Subject: RE: [Asterisk-Users] VoipBuster again




Iax.conf


register = username:[EMAIL PROTECTED]

Extensions.conf

exten =
_0.,2,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:\1\},\
60,r)

Good luck :) Sander

-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf
Ladyzhenskii
Verzonden: zaterdag 10 september 2005 13:57
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] VoipBuster again

Hi, all

I am still battling to connect * and voipbuster.

What protocol does it use? Ethereal capture shows UDP traffic, but no SIP
or
IAX traffic when using their client.

VoipBuster client connects to connectionserver.voipbuster.com on port
2
for authentication. Call itself is placed on different server.

I have tried to connect using SIP and IAX and it seems that no
authentication is happening. (i was trying to use sip.voipbuster.com and
iax.voipbuster.com). Does anyone currently use asterisk with voipbuster?
If
so, can you you help me to set it up? I am really lost.

My setup is :
sip.conf

[voipbuster]
type=peer
insecure=very
host=sip.voipbuster.com
username=NAME
secret=SECRET
fromdomain=sip.voipbuster.com
realm=voipbuster.com


iax.conf:
[voipbuster]
type=peer
host=iax.voipbuster.com
username=NAME
secret=NAME
notransfer=yes
qualify=no

extensions.conf: (I use 0 to dial out to IAX and 8 to dial out SIP) exten
=
_0.,1,SetCallerID(CID Name CIDNUMBER) exten =
_0.,2,Dial,IAX2/voipbuster/00613${EXTEN:1}

exten = _8.,1,SetCallerID(CID Name CIDNUMBER) exten =
_8.,2,Dial,SIP/voipbuster/00613${EXTEN:1}


Thanks,
Rudolf

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Re: [Asterisk-Users] Fritz, mISDN, Help

2005-09-11 Thread Tzafrir Cohen

On Sun, Sep 11, 2005 at 11:29:00AM +0300, Konrads Smelkovs wrote:
 Haven't tried. The install scripts gets May's release and compiles
 with that. I think some serious porting will be nescessary. Anyone?

If you look at the tarball, you can see it has three patches (for
libpri, zaptel and asterisk). The zaptel one is small and should
probably applies. If not: I have a slightly modified version of it at
home. The libpri one doesn't, IIRC. I haven't checked the asterisk one.

Anybody working on this?

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[Asterisk-Users] Using RedirectAction with queues

2005-09-11 Thread Josip Gracin


Hello!

Is it legal to use RedirectAction to redirect a call that is waiting in 
a queue? 

The idea is to have an external application manage a queue via manager 
API.  The queue
would merely collect calls and play moh. 

I've tryed this already but asterisk sends SIP/Forbidden to the channel 
in queue,
after the channel has been redirected by RedirectAction, even though the 
response

to RedirectAction is Success.

I'll send more details if necessary, but I just wanted first make sure 
that this

is how it's supposed to be done.

Thanks in advance!

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RE: [Asterisk-Users] TE406p no interrupts

I modified wct4xxp.c and make clean; make linux26;
make install; reboot;
But the system is not rebooted.
Because the system is in remote office I will check it
next morning.
Could you let me know your linux version, * version
and motherboard?

Thank you Boris.

--- Boris Bakchiev [EMAIL PROTECTED] wrote:

 Well.
 Try this please (but only if you're running on the
 latest sources).
 Open wct4xxp.c sources and search for
 pci_module_init
 Replace it with pci_register_driver
 So the line should read:
 res = pci_register_driver(t4_driver);
 
 That allows you to get the card working on 2.6.13 in
 almost exactly the
 same setup as yours.
 
 One weird thing though. Do no use insmod
 ./wct4xxp.ko from zaptel
 directory as it will not work. Do a proper make
 install and then
 modprobe.
 
 
 This is just part of the fixes you might need to do.
 If you encounter a problem after span
 reconfiguration (ztcfg) let me
 know.
 
 If you get stuck.. let me know.
 
 Regards
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Jason Kim
 Sent: Sunday, September 11, 2005 8:14 PM
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] TE406p no interrupts
 
 I'm using FC3.
 
 uname -a
 -
 Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2
 15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux
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Re: [Asterisk-Users] Special handling of IAX circuit-busy vs busy

2005-09-11 Thread Kevin P. Fleming


[EMAIL PROTECTED] wrote:

Is there a way to change our dialplan to fail to PSTN in case Dial(*) 
reports circuit-busy (but not busy)?  I'd like to send to another part of 
extensions.conf, where we'd try Dial(Zap).  We're already using the n+101 
extension to handle the busy condition with the Busy() app.


Dial will report CONGESTION in this case, which is separate from BUSY. 
This should be available in the DIALSTATUS channel variable after Dial() 
returns.

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Re: [Asterisk-Users] Pass through of T.38

2005-09-11 Thread Kevin P. Fleming


Matthew Boehm wrote:

I almost had to change my pants when I saw a CVS update this morning 
adding T38 frame recognition to asterisk. I kept looking for the code 
that complimented this but haven't seen it yet. And there was no bug 
reference so I can't help test.


Interested parties can easily find the open bug in Mantis where this is 
being discussed. There was no bug reference in the CVS commit since it 
did not come from a patch... but T.38 passthrough is being worked on.

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Re: [Asterisk-Users] False Zap answer problem (Again)

2005-09-11 Thread Soner Tari

I've reported this issue as a bug, and learned that a workaround seems to be 
disabling callwaiting :(. Since callwaiting was quite helpful on my systems, 
I'm not happy with this.


Anyone interested can go to issue tracer and follow if a fix will be 
released soon (ID# 0005188).


I really tend to think that there is a bug in chan_zap.c, because this 
issue happens only when the first caller hangs up first, never the 
otherwise. I mean if the second caller hangs up first, the first caller 
continues as usual without any problems and reaches to the VoiceMail system 
as expected. Please see below, in this case there is no problem at all:


   -- Executing Dial(SIP/201-e848, ZAP/1|15|tr) in new stack
   -- Called 1
   -- Zap/1-1 is ringing
   -- Executing Dial(SIP/202-c185, ZAP/1|15|tr) in new stack
   -- Called 1
   -- Zap/1-2 is ringing
   -- Zap/1-1 is ringing
   -- CPE does not support Call Waiting Caller*ID.
   -- Hungup 'Zap/1-2'
 == Spawn extension (from-internal, 200, 1) exited non-zero on 
'SIP/202-c185'

   -- Zap/1-1 is ringing
   -- Zap/1-1 is ringing
   -- Nobody picked up in 15000 ms
   -- Hungup 'Zap/1-1'
   -- Executing VoiceMail(SIP/201-e848, [EMAIL PROTECTED]) in new stack
   etc.

Please see below example for the problem case where the second caller 
cannot reach at VoiceMail system, but hears ringing indefinately, thus the 
problem. This cannot be the expected behaviour.


I was suspicious about Call Waiting Caller*ID, so I disabled it, but 
nothing changes. Still, I am concerned about it, because the problem 
happens to the Call Waiting (i.e. the second) caller (I mean to the caller 
who causes the Call Waiting for the callee).


What's going on here? Can somebody try and comment please? Any ideas?
Thanks,
Soner


I can replicate this issue on my test system with 1x FXS module too:
   -- Executing Dial(SIP/202-ea85, ZAP/1|15|tr) in new stack
   -- Called 1
   -- Zap/1-1 is ringing
   -- Zap/1-1 is ringing
   -- Executing Dial(SIP/201-bf73, ZAP/1|15|tr) in new stack
   -- Called 1
   -- Zap/1-2 is ringing
   -- CPE does not support Call Waiting Caller*ID.
   -- Zap/1-1 is ringing
   -- Hungup 'Zap/1-1'
 == Spawn extension (from-internal, 200, 1) exited non-zero on 
'SIP/202-ea85'

   -- Zap/1-2 answered SIP/201-bf73

When the channels are internal as they are above, it looks like a 
nonissue and seems like an expected behaviour.


But if the call is from external (FXO lines), hangup detection using busy 
detect is affected, and my PSTN lines stay open and Asterisk keeps 
attempting to bridge the 2 channels (FXO and FXS).


Can somebody comment please? Should this be the expected behaviour of 
chan_zap?

Soner

I've been monitoring this problem for almost a month now. I realized 
that it is more extensive than what I described previously. I can very 
easily replicate this problem on every Zap channel. Following is the 
senario:


1. Call Zap/5 via say SIP/15 -
   Zap/5-1 created and starts to ring
2. Call Zap/5 via say SIP/21 -
   Zap/5-2 created and starts to ring
3. Hangup SIP/15 -
   Zap/5-1 hungup

Right after this point we have the problem (please see full log below 
for details):
Sep 10 19:22:41 VERBOSE[27367] logger.c: -- Zap/5-2 answered 
SIP/21-efcb


When in fact nobody is answering Zap/5-2 !!! And on SIP/21 I hear 
strange ringing tone indefinetly, untill I hangup SIP/21.


What the hell is going on here? I don't have any other problem, this 
system is in use for 1.5 month now (Users cannot notice it, because they 
hangup immediately).


Since I can replicate this problem with Zap/6 and Zap/7 also, I tend to 
think that this is not specific to any FXS module. But, of course, it 
could be the TDM PCI card itself. Could this be a bug in chan_zap.c?


Can somebody please confirm that using the same senario this only 
happens on my system with my TDM card, so I don't file a bug report?


Please find below the relevant sections of full log and my previous 
post,

Thanks,
Soner

Sep 10 19:22:33 DEBUG[27367] chan_sip.c: Checking SIP call limits for 
device 15
Sep 10 19:22:33 DEBUG[27367] chan_sip.c: build_route: Contact hop: 
sip:[EMAIL PROTECTED]:51926
Sep 10 19:22:33 VERBOSE[27367] logger.c: -- Executing 
Macro(SIP/15-f784, ichatarama|Zap/5|10) in new stack
Sep 10 19:22:33 VERBOSE[27367] logger.c: -- Executing 
GotoIf(SIP/15-f784, =1?200) in new stack

Sep 10 19:22:33 DEBUG[27367] pbx.c: Not taking any branch
Sep 10 19:22:33 VERBOSE[27367] logger.c: -- Executing 
Dial(SIP/15-f784, Zap/5|24|rTtWw) in new stack

Sep 10 19:22:33 VERBOSE[27367] logger.c: -- Called 5
Sep 10 19:22:33 VERBOSE[27367] logger.c: -- Zap/5-1 is ringing
Sep 10 19:22:34 DEBUG[27367] chan_sip.c: Setting NAT on RTP to 524288
Sep 10 19:22:35 DEBUG[27367] chan_sip.c: Setting NAT on RTP to 524288
Sep 10 19:22:35 DEBUG[27367] chan_sip.c: Checking SIP call limits for 
device 21
Sep 10 19:22:35 DEBUG[27367] chan_sip.c: build_route: Contact hop: 
sip:[EMAIL PROTECTED]:46209
Sep 10 19:22:35

[Asterisk-Users] rotate * log file?

2005-09-11 Thread Rich Adamson


Running fc3 with current cvs-head...

Is there a nice way to rotate the /var/log/asterisk/messages file without
shutting down asterisk?

I'm currently rotating the log files via cron, however my script requires
asterisk to be shut down, which also kills any outstanding cli sessions
(eg, asterisk -rv). Would like to rotate the files without killing
the cli session. Any reasonable way to accomplish this?

Rich


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Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

2005-09-11 Thread Mike M

On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote:
 
 Mark Phillips wrote:
 
 The suggestion that I have is for various areas to have dedicated civil 
 emergency com units with strategic reserves of fuel (3-4 weeks worth), 
 battery backups, etc.  These units would have links (fiber, microwave, 
 and/or satellite, better to pick 2 of 3) to areas outside expected 
 disaster zones.  Asterisk could then run across these links.  (Sattelite 
 links would best be POTS-type).
 
 The point is to a disaster-tolerant communications infrastructure which 
 could then be used to to provide additional communications services to 
 the relief workers.  With various point to point wireless capabilities, 
 it might be possible to use them to provide cell service to relief 
 workers etc through the installation of GSM microcells (which could be 
 brought in after the fact).
 
 See where I am going?

Great suggestions but these are out of the realm of what a community of
individuals can do.  I'm thinking about what I as an individual am
capable of.

-- 
Mike
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Re: [Asterisk-Users] rotate * log file?

2005-09-11 Thread Matt Riddell

Rich Adamson wrote:
 Running fc3 with current cvs-head...
 
 Is there a nice way to rotate the /var/log/asterisk/messages file without
 shutting down asterisk?
 
 I'm currently rotating the log files via cron, however my script requires
 asterisk to be shut down, which also kills any outstanding cli sessions
 (eg, asterisk -rv). Would like to rotate the files without killing
 the cli session. Any reasonable way to accomplish this?

In the Asterisk console type:

logger rotate

Surely this would have come up in a google search.  (You can find google at
http://www.google.com - it's a search engine)

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] AGI + Ruby

2005-09-11 Thread joe heitzeberg

Hi,

We have created RAGI (Ruby Asterisk Gateway Interface) for the open
source community so that Ruby and Ruby on Rails can be used to easily
and effeciently create Asterisk-based applications.  Examples:  IVR,
call center apps, Asterisk management consoles, etc.

RAGI includes a set of objects to interface over AGI to Asterisk for
handling inbound calls and outbound dialing, and includes a server
component, documentation and a sample apps to get you going quickly.

Please see: http://ragi.sourceforge.net/

The prelimenary release is available now on
https://sourceforge.net/projects/ragi

We welcome input and development participation in the effort.


thanks,
Joe Heitzeberg
SNAPVINE



On 8/24/05, Innocent Evil [EMAIL PROTECTED] wrote:
 I would like to write AGI script in Ruby
 Would anybody please show me right direction..
 
 
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Re: [Asterisk-Users] rotate * log file?

2005-09-11 Thread Tzafrir Cohen

On Sun, Sep 11, 2005 at 10:26:38AM -0600, Rich Adamson wrote:
 
 Running fc3 with current cvs-head...
 
 Is there a nice way to rotate the /var/log/asterisk/messages file without
 shutting down asterisk?
 
 I'm currently rotating the log files via cron, however my script requires
 asterisk to be shut down, which also kills any outstanding cli sessions
 (eg, asterisk -rv). Would like to rotate the files without killing
 the cli session. Any reasonable way to accomplish this?

Here's what Debian installs at /etc/logrotate.d/asterisk:

/var/log/asterisk/cdr-csv/Master.csv /var/log/asterisk/debug 
/var/log/asterisk/event_log /var/log/asterisk/messages {
weekly
missingok
rotate 4
sharedscripts
postrotate
/usr/sbin/invoke-rc.d asterisk logger-reload
endscript
}



Naturally the post-rotate script may differ on your system. logger-reload
is a glorified 'asterisk -rx logger reload' . logrotate is a standard part 
of most linux distros.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

2005-09-11 Thread Derek Whitten

On Sun, 2005-09-11 at 08:44, Mike M wrote:
 On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote:
  
  Mark Phillips wrote:
  
  The suggestion that I have is for various areas to have dedicated civil 
  emergency com units with strategic reserves of fuel (3-4 weeks worth), 
  battery backups, etc.  These units would have links (fiber, microwave, 
  and/or satellite, better to pick 2 of 3) to areas outside expected 
  disaster zones.  Asterisk could then run across these links.  (Sattelite 
  links would best be POTS-type).
  
  The point is to a disaster-tolerant communications infrastructure which 
  could then be used to to provide additional communications services to 
  the relief workers.  With various point to point wireless capabilities, 
  it might be possible to use them to provide cell service to relief 
  workers etc through the installation of GSM microcells (which could be 
  brought in after the fact).
  
  See where I am going?
 
 Great suggestions but these are out of the realm of what a community of
 individuals can do.  I'm thinking about what I as an individual am
 capable of.

These are great suggestions and I believe that it IS in the realm of
'what a community of individuals can do' .. It just depends on the
community of individuals involved with the project..  


:-0



my 0.02


-- 
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Version: 3.1
GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y 
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[Asterisk-Users] ruby-agi 0.0.2 released

2005-09-11 Thread Mohammad Khan


Hello,

I have released Ruby Asterisk Gateway Interface (ruby-agi) v0.0.2b.
Any feedback, bug report, suggession, feature request is most welcome.

ruby-agi homepage:
http://www.rubyforge.org/projects/ruby-agi/

Download ruby-agi v0.0.2b here:
http://rubyforge.org/frs/download.php/5965/ruby-agi_v0.0.2b.tgz


Thanks,
Mohammad Khan

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[Asterisk-Users] cdr_addon_mysql.so pb

2005-09-11 Thread alexandre zhang

hi 

I load cdr_addon_mysql.so without error 

configuration of cdr_mysql.conf 

[general]dbhost = localhostdbname = rechargedbuser = rootdbpass = astdbport = 3306dbsock = /var/lib/mysql/mysql.sock

But, Iget nothingin the table of cdrof my database.


Somebody have an idea ? Thanks you for your help

best regards

		DO YOU YAHOO!? 
 
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Re: [Asterisk-Users] rotate * log file?

2005-09-11 Thread Rich Adamson

 Rich Adamson wrote:
  Running fc3 with current cvs-head...
  
  Is there a nice way to rotate the /var/log/asterisk/messages file without
  shutting down asterisk?
  
  I'm currently rotating the log files via cron, however my script requires
  asterisk to be shut down, which also kills any outstanding cli sessions
  (eg, asterisk -rv). Would like to rotate the files without killing
  the cli session. Any reasonable way to accomplish this?
 
 In the Asterisk console type:
 
 logger rotate
 
 Surely this would have come up in a google search.  (You can find google at
 http://www.google.com - it's a search engine)

Ops, my bad; must have been the lack of coffee as I didn't even
attempt the most basic of searches. I actually started digging through
this thinking I had bumped into a problem from yesterday's cvs head
update, but then realized it resulted from our own limited script.
Never gave it a thought that * already had it.

Thanks


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RE: [Asterisk-Users] cdr_addon_mysql.so pb

2005-09-11 Thread Jonathan k. Creasy









Did you confirm that cdr_addon_mysql is
indeed built and loading? I had missed that it wasn’t being built (I didn’t
have mysql-devel installed) when I first tried to do that a few months ago. 



-Jonathan









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of alexandre zhang
Sent: Sunday, September 11, 2005 2:11 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
cdr_addon_mysql.so pb







hi 











I
load cdr_addon_mysql.so without error 











configuration
of cdr_mysql.conf 











[general]
dbhost = localhost
dbname = recharge
dbuser = root
dbpass = ast
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock











But,
Iget
nothingin
the table of cdrof
my database.

















Somebody
have an idea ? Thanks you for your help











best
regards















DO
YOU YAHOO!?
雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 






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Re: [Asterisk-Users] cdr_addon_mysql.so pb

2005-09-11 Thread Matthew Boehm

Run the command cdr mysql status from asterisk CLI. What does that say? If
it says command not found then the module is not loaded.

-Matthew


 From: alexandre zhang [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST)
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] cdr_addon_mysql.so pb
 
 hi 
  
 I load cdr_addon_mysql.so without error
  
 configuration  of cdr_mysql.conf
  
 [general]
 dbhost = localhost
 dbname = recharge
 dbuser = root
 dbpass = ast
 dbport =  3306
 dbsock = /var/lib/mysql/mysql.sock
 
  
 But, I get nothing in the table of cdr of my database.
  
  
 Somebody have an idea ? Thanks you for your help
  
 best regards
  
 
 
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[Asterisk-Users] Australian Dial tone TDM400P

2005-09-11 Thread Asterisk Sales

hello asterisk users,
i an using asterisk cvs 1.0.9 in a pIII 733mhz 256MB RAMredhat 9.
i have a TDM400P with 2FXO and 2FXS modules. in my fxs i want to get australian dial tone and for all asterisk operation i want to use australian tones. by default it is US. to change this i have edited following files:


/etc/zaptel.conf (loadzone = au, defaultzone= au)
/etc/asterisk/indications.conf (country= au)

but for all kind of signals and tones i stillget US tones. i restarted asterisk and run it #asterisk -vc
no luck.
please help

best regards
shaon
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回复： RE: [Asterisk-Users] cdr_addon_my sql.so pb

2005-09-11 Thread alexandre zhang


thanks u for ur help

cdr_addon_mysql is built without error. I got the cdr_addon_mysql.so 
I input cmd 'show modules' under CLI. I saw cdr_addon_mysql.so .
I built also res_config_mysql.soat same time and it works fine.

Thanks
"Jonathan k. Creasy" [EMAIL PROTECTED] 写道：









Did you confirm that cdr_addon_mysql is indeed built and loading? I had missed that it wasn’t being built (I didn’t have mysql-devel installed) when I first tried to do that a few months ago. 

-Jonathan




From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of alexandre zhangSent: Sunday, September 11, 2005 2:11 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] cdr_addon_mysql.so pb


hi 



I load cdr_addon_mysql.so without error 



configuration of cdr_mysql.conf 



[general]dbhost = localhostdbname = rechargedbuser = rootdbpass = astdbport = 3306dbsock = /var/lib/mysql/mysql.sock



But, Iget nothingin the table of cdrof my database.





Somebody have an idea ? Thanks you for your help



best regards





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Re: [Asterisk-Users] cdr_addon_mysql.so pb

2005-09-11 Thread JOAO CARLOS MOURA


I use the module that this in the [EMAIL PROTECTED] and functions very well.

[]'s
jmoura

- Original Message - 
From: Jonathan k. Creasy

To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, September 11, 2005 15:00
Subject: RE: [Asterisk-Users] cdr_addon_mysql.so pb


Did you confirm that cdr_addon_mysql is indeed built and loading? I had 
missed that it wasn’t being built (I didn’t have mysql-devel installed) 
when I first tried to do that a few months ago.


-Jonathan



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of alexandre 
zhang

Sent: Sunday, September 11, 2005 2:11 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] cdr_addon_mysql.so pb

hi

I load cdr_addon_mysql.so without error

configuration  of cdr_mysql.conf

[general]
dbhost = localhost
dbname = recharge
dbuser = root
dbpass = ast
dbport =  3306
dbsock = /var/lib/mysql/mysql.sock

But, I get nothing in the table of cdr of my database.


Somebody have an idea ? Thanks you for your help

best regards




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Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

2005-09-11 Thread Chris Travers


Michael D Schelin wrote:

The two best forms of communications in a real disaster and one always 
has been is #1 Ham radio. and #2 satellite telephone. Ham radio is 
global and has proven time and time again to be the most reliable when 
the infrastructer has been damaged.  The U.S government is the biggest 
user of satellite telephones which is also becoming a valuable tool 
again when the communications infrastructure is down.  It would be 
nice If Asterisk could be used but in this case but it's useless.  
People are displaced and most of the communications infrastructure for 
the city is unusable.  I don't mean all of the telco's systems. It's 
the flood that wiped out  most home and business systems.  For us, The 
best thing that a provider can do is to have redundant servers in 
different cities.  This should remind us all how fragile our lives are.


While I agree with your points, I think I was thinking along different 
lines.  Your points are useful particularly for mobile units.  This is 
important because you have to have some form of mobile communications 
when you are doing disaster relief.  I am not saying that my suggestion 
would relieve the need for Ham radio and Satellite telephone.  But 
rather that this would allow you to do relatively quick infrastructure 
building to fixed locations thus freeing up Ham operators to do what 
they need to do-- offer mobile communications.  The idea here would be 
that shelters, etc. could then use various line-of-site wireleass 
connections to set up Asterisk and that these would not have to be moved 
frequently.  Yes, it takes more electricity, but remember what I said 
about strategic reserves of fuel for generators?


I was largely reacting to Mark Phillips' point about Ham radios being in 
short supply in any sort of disaster.  The point is not to replace ham 
radio but rather to maximize the potential of what can be done with the 
existing number of operators.


Best Wishes,
Chris Travers
Metatron Technology Consulting
begin:vcard
fn:Chris Travers
n:Travers;Chris
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
version:2.1
end:vcard

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Re: [Asterisk-Users] TE110P reset

2005-09-11 Thread JOAO CARLOS MOURA

Thank you for all
Sorry my English
Jmoura

- Original Message - 
From: Jason Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Saturday, September 10, 2005 21:40
Subject: RE: [Asterisk-Users] TE110P reset

You are correct. I did not expand completely and stand corrected. An
additional note...we have some Dialogic cards (not associated with *) that
do the same thing on PRI.

Question - is it somewhat standard to have b chans restart on PRI circuits
when not explicitly configured to NOT reset?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Saturday, September 10, 2005 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] TE110P reset

On Saturday 10 September 2005 19:40, Jason Walker wrote:

PRI channels will reset when not in use throughout the day. A reset on
a channel should not happen when that channel is in use. This happens
all the time on my PRI circuits (TE110P and TE410P). From what I
gather, it's somewhat like a handshake for the D chan between the cpe and

net sides.

Not exactly.  Digium's replicating the B channel resets someone noted in a
particular situation.  It's not required, but it shouldn't hurt.  If it's
causing trouble you can turn it off with resetinterval=0 in your
zapata.conf.

-A.
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[Asterisk-Users] David Choo/eServices/eSpore is overseas

2005-09-11 Thread David Choo


I will be out of the office starting  12/09/2005 and will not return until
16/09/2005.

Dear Sir / Mdm,

I'm currently on course and are not in office.

During this period of time, I have minimal access to internet and email
cccess. As such, I might not be able to reply to your queries promptly. I
apologise for the inconvenience caused.

In the meantime, for any technical assitance, please contact the Espore
Technical Support Hotline at +65-68422725 and select option 2.

However, during this period of time, I'm still contacted via my Mobile
Phone. Please feel free to contact me should you feel necessary.

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回复： Re: [Asterisk-Users] cdr_addon_my sql.so pb

2005-09-11 Thread alexandre zhang

thanks for ur help

Run the command "cdr mysql status" 
I got the following msg
'No such command 'cdr mysql' (type 'help' for help)'

But, I run 'show modules'
cdr_addon_mysql.so is in the list

Do u have an idea about it ?

Thanks
Matthew Boehm [EMAIL PROTECTED] 写道：
Run the command "cdr mysql status" from asterisk CLI. What does that say? Ifit says command not found then the module is not loaded.-Matthew From: alexandre zhang <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion  Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST) To:  Subject: [Asterisk-Users] cdr_addon_mysql.so pb  hi   I load cdr_addon_mysql.so without error  configuration of cdr_mysql.conf  [general] dbhost = localhost dbname = recharge dbuser = root dbpass = ast dbport = 3306 dbsock = /var/lib/mysql/mysql.sock   But, I get nothing in the table of cdr of my database. <
 BR>
  Somebody have an idea ? Thanks you for your help  best regards- DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --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___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Make asterisk call out

2005-09-11 Thread Samy Antoun

Andreas,

You may like to take a look at
http://mundy.org/blog/index.php
The part of Call Out feature is near half page, the
paragraph heading is Phone Home (with ET Image !!!)

I tried it and it work great.


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RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM




Time and time again, emergency action drills take place in 
cities to target where their weaknesses are in "crisis" handling. Usually 
they involve planes crashing or explosions (mock of course). Obviously 
they were never prepared for this sort of disaster in their recovery plan. 
I've participated in a few ARES/RACES drills and have to say that much could be 
done to improve upon the "HAM" infrastructure.

Most of the time, communications is coordinated through 1 
repeater system. When this repeater goes down, of course people would 
switch comms to another but in a case like this, where all the repeater systems 
go down except for maybe one, there needs to be a better 
plan.

In Amateur Satellite Service, these orbiting "Repeaters" 
employ a system called RUDAK where a chunk of spectrum is repeated. 
Obviously this isn't feasible in terrestrial repeaters but they dohave the 
ability to turn off radios and switch bands at will depending on operating 
conditions. With software controlled radio and Asterisk, the repeater 
system could be made to be more resilient to disaster by linking to other 
repeater systems via radio where it could connect outward. 


If you figure the overhead of a repeater's transmitter and 
receiver plus the controller, replaceing the controller with an asterisk based 
unit (integration) would make more sense as it would give the repeater system 
much more capabilities in the same footprint and power. Additionally, 
these repeater systems are located on hilltops with other radio systems so they 
should have emergency power available (if you've ever been to a hilltop repeater 
site, you'll know what I mean). 

I think the biggest thing that hurts ham radio's ability to 
react to a crisis is the lack of equipment and operators. Most of the 
traffic we pass is "Health and Welfare" with "Logistics" being the second to 
it. What defeats this is that in a disaster where local/high band long 
haul capabilities are diminished, is simply the one repeater that is functional 
because everything is squeezed onto one VHF/UHF repeater.

Where I could see thing being improved? Installation 
of 802.11b/g WLAN under Part 97. It would allow for more users into the 
system, there are less hardware and power components and allows the system to be 
dynamically configured. Asterisk could play a huge role then as it's made 
for IP based traffic and could re-route in a split second.

-Don



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D 
SchelinSent: Saturday, September 10, 2005 10:20 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] civil emergency comms: Asterisk + HAM
The two best forms of communications in a real disaster and one 
always has been is #1 Ham radio. and #2 satellite telephone. Ham radio is global 
and has proven time and time again to be the most reliable when the 
infrastructer has been damaged. The U.S government is the biggest user of 
satellite telephones which is also becoming a valuable tool again when the 
communications infrastructure is down. It would be nice If Asterisk could 
be used but in this case but it's useless. People are displaced and most 
of the communications infrastructure for the city is unusable. I don't 
mean all of the telco's systems. It's the flood that wiped out most home 
and business systems. For us, The best thing that a provider can do is to 
have redundant servers in different cities. This should remind us all how 
fragile our lives are. Chris Travers wrote:
Mark 
  Phillips wrote: 
  Hold on here folks, I'm guessing that the 
original poster of this thread isn't a member of his local RAyNet team. 
Whilst I don't profess to be an expert at this I have been doing 
emergency radio for quite some time and have seen service at the Lockerbie 
bombing, Docklands bomb, Ground Zero (I'm sure I'm a terrorist target y'know 
- they seem to follow me everywhere) and soon I'll be in Louisiana. 
In all of these events the KISS principle must and does prevail. We 
need a system that is a simple and energy efficient as possible. 
  
  Building a network of * servers and Wi-Fi links 
is all very well but how are you going to power them? 
  These are excellent points. I have a few 
  interesting suggestions here The first is that the only obstacle to 
  any sort of longer-range point to point line is merely power. This is 
  true whether you are talking HAM or fiberoptics. Note that if you have 
  the power, it would take disruption of the physical line to disrupt a fiber 
  line. Note that DirectNIC in New Orleans remained operational without 
  *any* downtime or loss of connectivity with the rest of the world. The 
  suggestion that I have is for various areas to have dedicated civil emergency 
  com units with strategic reserves of fuel (3-4 weeks worth), battery backups, 
  etc. These units would have links (fiber, microwave, and/or satellite, 
  better to

[Asterisk-Users] H323 with asterisk-ooh323c

2005-09-11 Thread Adam Rybak

Hello,

i have succesfully compiled and installed newest channel driver ooh323c with
asterisk CVS-HEAD.
I have small problem - when the asterisk logins to the GnuGK its shchown as
unknown type:
RCF|195.214.XXX.XXX:1720|ASTERIX2:h323_ID|unknown|9681_endp
Sun, 11 Sep 2005 22:34:34 +0200 C(0/0/0)  1
and when im seting prefixes for routing in gnugk.ini this not working.
If i use oh323 channel (0.7.1pre) this logins as gateway:
RCF|195.214.XXX.XXX:1720|ASTERIX:h323_ID|gateway|7452_endp
Sun, 11 Sep 2005 22:29:51 +0200 C(0/0/6)  2
Prefixes: 881,871

how to change that ooh323c login as gateway type?
I need send traffic from H.323 network to the asterisk.

How to configuree h323.conf?
My h323 conf is:
[general]
port=1720
bindaddr=195.214.XXX.XXX
faststart=yes
h245tunneling=no
h323id=ASTERIX2
gatekeeper = 195.214.XXX.XXX
logfile=/var/log/asterisk/h323_log
context=in
disallow=all
allow=gsm
allow=ilbc
dtmfmode=rfc2833
[incoming]
type=user
context=in
allow=all
prefix=*


Pozdrawiam,
Adam Rybak
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Re: 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb

2005-09-11 Thread Matthew Boehm

cytrex2*CLI cdr mysql status
Connected to [EMAIL PROTECTED], port 3306 using table cdr for 3 days, 18
hours, 4 minutes, 3 seconds.
  Wrote 62028 records since last restart.

That is what you should see.

-Matthew

 From: alexandre zhang [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Mon, 12 Sep 2005 04:07:28 +0800 (CST)
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb
 
 thanks for ur help
  
 Run the command cdr mysql status
 I got the following msg
 ' No such command 'cdr mysql' (type 'help' for help)'
  
 But, I run ' show modules'
 cdr_addon_mysql.so  is in the list
  
 Do u have an idea  about it ?
  
 Thanks
 
 
 
 Matthew Boehm [EMAIL PROTECTED] 写道：
 Run the command cdr mysql status from asterisk CLI. What does that say? If
 it says command not found then the module is not loaded.
 
 -Matthew
 
 
 From: alexandre zhang
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 
 Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST)
 To: 
 Subject: [Asterisk-Users] cdr_addon_mysql.so pb
 
 hi 
 
 I load cdr_addon_mysql.so without error
 
 configuration of cdr_mysql.conf
 
 [general]
 dbhost = localhost
 dbname = recharge
 dbuser = root
 dbpass = ast
 dbport = 3306
 dbsock = /var/lib/mysql/mysql.sock
 
 
 But, I get nothing in the table of cdr of my database.
 
 
 Somebody have an idea ? Thanks you for your help
 
 best regards
 
 
 
 -
 DO YOU YAHOO!?
 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

2005-09-11 Thread Michael D Schelin





Don, I agree with you on many fronts. I come from a radio background
and here in southern cal unless we fall into the sea nothing will take
out all of the communications here including ham because we are not in
low lying flat land and were too diversified, over 150 miles and as
many mountain top sites. 
BUT,
let me tell you about how bad the southern CA. radio site owners are
becoming. We had a 4 day outage at a very large site where one of my
radios is located. None of them care anymore about backup power. This
happened this past week. We took up our own Generator because the site
owner (a national site company) won't maintain an old one. My friend
(a microwave isp ) fixed the site owners by adding oil and a new
battery. That will take us out!


Don Fanning wrote:

  
  
  
  Time and time again, emergency
action drills take place in cities to target where their weaknesses are
in "crisis" handling. Usually they involve planes crashing or
explosions (mock of course). Obviously they were never prepared for
this sort of disaster in their recovery plan. I've participated in a
few ARES/RACES drills and have to say that much could be done to
improve upon the "HAM" infrastructure.
  
  Most of the time, communications
is coordinated through 1 repeater system. When this repeater goes
down, of course people would switch comms to another but in a case like
this, where all the repeater systems go down except for maybe one,
there needs to be a better plan.
  
  In Amateur Satellite Service,
these orbiting "Repeaters" employ a system called RUDAK where a chunk
of spectrum is repeated. Obviously this isn't feasible in terrestrial
repeaters but they dohave the ability to turn off radios and switch
bands at will depending on operating conditions. With software
controlled radio and Asterisk, the repeater system could be made to be
more resilient to disaster by linking to other repeater systems via
radio where it could connect outward. 
  
  If you figure the overhead of a
repeater's transmitter and receiver plus the controller, replaceing the
controller with an asterisk based unit (integration) would make more
sense as it would give the repeater system much more capabilities in
the same footprint and power. Additionally, these repeater systems are
located on hilltops with other radio systems so they should have
emergency power available (if you've ever been to a hilltop repeater
site, you'll know what I mean). 
  
  I think the biggest thing that
hurts ham radio's ability to react to a crisis is the lack of equipment
and operators. Most of the traffic we pass is "Health and Welfare"
with "Logistics" being the second to it. What defeats this is that in
a disaster where local/high band long haul capabilities are diminished,
is simply the one repeater that is functional because everything is
squeezed onto one VHF/UHF repeater.
  
  Where I could see thing being
improved? Installation of 802.11b/g WLAN under Part 97. It would
allow for more users into the system, there are less hardware and power
components and allows the system to be dynamically configured.
Asterisk could play a huge role then as it's made for IP based traffic
and could re-route in a split second.
  
  -Don
  
  
  
  From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Michael
D Schelin
  Sent: Saturday, September 10, 2005 10:20 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] civil emergency comms: Asterisk
+ HAM
  
  
The two best forms of communications in a real disaster and one always
has been is #1 Ham radio. and #2 satellite telephone. Ham radio is
global and has proven time and time again to be the most reliable when
the infrastructer has been damaged. The U.S government is the biggest
user of satellite telephones which is also becoming a valuable tool
again when the communications infrastructure is down. It would be nice
If Asterisk could be used but in this case but it's useless. People
are displaced and most of the communications infrastructure for the
city is unusable. I don't mean all of the telco's systems. It's the
flood that wiped out most home and business systems. For us, The best
thing that a provider can do is to have redundant servers in different
cities. This should remind us all how fragile our lives are. 
  
Chris Travers wrote:
  
Mark Phillips wrote: 

Hold on here folks, 
  
I'm guessing that the original poster of this thread isn't a member of
his local RAyNet team. 
  
Whilst I don't profess to be an expert at this I have been doing
emergency radio for quite some time and have seen service at the
Lockerbie bombing, Docklands bomb, Ground Zero (I'm sure I'm a
terrorist target y'know - they seem to follow me everywhere) and soon
I'll be in Louisiana. 
  
In all of these events the KISS principle must and does prevail. We
need a system that is a simple and energy efficient as possible. 



Building a network of * servers and Wi-Fi links

[Asterisk-Users] Call Waiting Tracking?

2005-09-11 Thread Nathan E. Pralle

Hi all.

Searched the archives but couldn't find anything on this:

I want to track 2nd incoming calls on a single line but don't want to pass the 
Call Waiting pips along to the engaged user.  IE:  I want Asterisk to detect 
that CW is currently being transmitted on the line, and track it, but not 
pass it on.

Is there a way to do this?

TIA,
Nathan



-- 

Interesting things abide:
http://www.nathanpralle.com

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Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

2005-09-11 Thread Steve Totaro

Just a shot in the dark here.

I bought this unit http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5792377951rd=1sspagename=STRK%3AMEWN%3AITrd=1a
couple months ago hoping to connect it to an * system for experimentation.
I am a HAM n00b. I can found no documentation on this unit anywhere.
Does anyone know where to start?

I joined a local HAM club but have not had any time
to go and pick brains. I am afraid to really even plug it in until I know
what I am doing and have a call sign and everything so the FCC does't kick in my
door. I did plug it in for a minute and there were no lights or anything
so I not even sure it works.

Anyone have any links or ideas?

Thanks,
Steve

- Original Message -
From:
Don Fanning

To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, September 11, 2005 1:37
PM
Subject: RE: [Asterisk-Users] civil
emergency comms: Asterisk + HAM

Time and time again, emergency action drills take place
in cities to target where their weaknesses are in "crisis" handling.
Usually they involve planes crashing or explosions (mock of course).
Obviously they were never prepared for this sort of disaster in their recovery
plan. I've participated in a few ARES/RACES drills and have to say that
much could be done to improve upon the "HAM"
infrastructure.

Most of the time, communications is coordinated through 1
repeater system. When this repeater goes down, of course people would
switch comms to another but in a case like this, where all the repeater
systems go down except for maybe one, there needs to be a better
plan.

In Amateur Satellite Service, these orbiting "Repeaters"
employ a system called RUDAK where a chunk of spectrum is repeated.
Obviously this isn't feasible in terrestrial repeaters but they dohave
the ability to turn off radios and switch bands at will depending on operating
conditions. With software controlled radio and Asterisk, the repeater
system could be made to be more resilient to disaster by linking to other
repeater systems via radio where it could connect outward.

If you figure the overhead of a repeater's transmitter
and receiver plus the controller, replaceing the controller with an asterisk
based unit (integration) would make more sense as it would give the repeater
system much more capabilities in the same footprint and power.
Additionally, these repeater systems are located on hilltops with other radio
systems so they should have emergency power available (if you've ever been to
a hilltop repeater site, you'll know what I mean).

I think the biggest thing that hurts ham radio's ability
to react to a crisis is the lack of equipment and operators. Most of the
traffic we pass is "Health and Welfare" with "Logistics" being the second to
it. What defeats this is that in a disaster where local/high band long
haul capabilities are diminished, is simply the one repeater that is
functional because everything is squeezed onto one VHF/UHF
repeater.

Where I could see thing being improved?
Installation of 802.11b/g WLAN under Part 97. It would allow for more
users into the system, there are less hardware and power components and allows
the system to be dynamically configured. Asterisk could play a huge role
then as it's made for IP based traffic and could re-route in a split
second.

-Don

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
SchelinSent: Saturday, September 10, 2005 10:20 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] civil emergency comms: Asterisk + HAM
The two best forms of communications in a real disaster and one
always has been is #1 Ham radio. and #2 satellite telephone. Ham radio is
global and has proven time and time again to be the most reliable when the
infrastructer has been damaged. The U.S government is the biggest user
of satellite telephones which is also becoming a valuable tool again when the
communications infrastructure is down. It would be nice If Asterisk
could be used but in this case but it's useless. People are displaced
and most of the communications infrastructure for the city is unusable.
I don't mean all of the telco's systems. It's the flood that wiped out
most home and business systems. For us, The best thing that a provider
can do is to have redundant servers in different cities. This should
remind us all how fragile our lives are. Chris Travers
wrote:
Mark
Phillips wrote:
Hold on here folks, I'm guessing that the
original poster of this thread isn't a member of his local RAyNet team.
Whilst I don't profess to be an expert at this I have been doing
emergency radio for quite some time and have seen service at the Lockerbie
bombing, Docklands bomb, Ground Zero (I'm sure I'm a

Re: [Asterisk-Users] TE110P reset

2005-09-11 Thread Steve Totaro

Is there a situtation where it could hurt to turn off the reset?  I have
never had a problem with it cutting off a call or anything but it is
somewhat concerning to see it in the console all the time.

Thanks,
Steve

- Original Message - 
From: JOAO CARLOS MOURA [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, September 11, 2005 2:05 PM
Subject: Re: [Asterisk-Users] TE110P reset

 Thank you for all
 Sorry my English
 Jmoura

 - Original Message - 
 From: Jason Walker [EMAIL PROTECTED]
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Sent: Saturday, September 10, 2005 21:40
 Subject: RE: [Asterisk-Users] TE110P reset

  You are correct. I did not expand completely and stand corrected. An
  additional note...we have some Dialogic cards (not associated with *)
that
  do the same thing on PRI.

  Question - is it somewhat standard to have b chans restart on PRI
circuits
  when not explicitly configured to NOT reset?

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
  Kohlsmith
  Sent: Saturday, September 10, 2005 5:00 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [Asterisk-Users] TE110P reset

  On Saturday 10 September 2005 19:40, Jason Walker wrote:
  PRI channels will reset when not in use throughout the day. A reset on
  a channel should not happen when that channel is in use. This happens
  all the time on my PRI circuits (TE110P and TE410P). From what I
  gather, it's somewhat like a handshake for the D chan between the cpe
and
  net sides.

  Not exactly.  Digium's replicating the B channel resets someone noted in
a
  particular situation.  It's not required, but it shouldn't hurt.  If
it's
  causing trouble you can turn it off with resetinterval=0 in your
  zapata.conf.

  -A.
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Re: [Asterisk-Users] Make asterisk call out

2005-09-11 Thread Darren Wiebe

I've got started coding something similar to this.  It will have a web 
interface and a little for reporting abilities.  How soon do you need 
this?  What I have will be open source when it is done but I've not been 
rushing at all.



Darren Wiebe
[EMAIL PROTECTED]

Brian Roy wrote:




On 9/11/05, *Andreas Moroder* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Is it possible to use asterisk to call automatically a list of number.

 
 
Yes, it's possible. It will require a little effort to do some 
scripting, but not much. They key to making Asterisk call out, will be 
using the call files.
 
Here is the wiki page 
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
 
With a little effort you can do this yourself, or request someone 
write something for you on the -biz list. Someone would do it for a 
few bucks.
 
-Brian
 
 

 




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[Asterisk-Users] Presence Fully Supported?

2005-09-11 Thread Trevor Peirce

I've seen lots about presence and Polycom phones recently. I've got  one 
here for evaluation but noticed other phones only seem to appear busy 
when they initiate a call. If they receive a call, they still show as 
available.


Is this a config problem on my part, or is that as far as presence is 
working right now?


Thanks!
Trev
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RE: [Asterisk-Users] TE406p no interrupts

You should have just done this:
rmmod wct4xxp
rmmod zaptel
modprobe wct4xxp

It will do the same thing

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jason Kim
 Sent: Monday, 12 September 2005 00:34
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] TE406p no interrupts
 
 I modified wct4xxp.c and make clean; make linux26;
 make install; reboot;
 But the system is not rebooted.
 Because the system is in remote office I will check it
 next morning.
 Could you let me know your linux version, * version
 and motherboard?
 
 Thank you Boris.
 
 --- Boris Bakchiev [EMAIL PROTECTED] wrote:
 
  Well.
  Try this please (but only if you're running on the
  latest sources).
  Open wct4xxp.c sources and search for
  pci_module_init
  Replace it with pci_register_driver
  So the line should read:
  res = pci_register_driver(t4_driver);
 
  That allows you to get the card working on 2.6.13 in
  almost exactly the
  same setup as yours.
 
  One weird thing though. Do no use insmod
  ./wct4xxp.ko from zaptel
  directory as it will not work. Do a proper make
  install and then
  modprobe.
 
 
  This is just part of the fixes you might need to do.
  If you encounter a problem after span
  reconfiguration (ztcfg) let me
  know.
 
  If you get stuck.. let me know.
 
  Regards
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On
  Behalf Of Jason Kim
  Sent: Sunday, September 11, 2005 8:14 PM
  To: asterisk-users@lists.digium.com
  Subject: RE: [Asterisk-Users] TE406p no interrupts
 
  I'm using FC3.
 
  uname -a
  -
  Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2
  15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux
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 http://lists.digium.com/mailman/listinfo/asterisk-users
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
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Re: [Asterisk-Users] Presence Fully Supported?

2005-09-11 Thread Paul Hales

The latest CVS versions support Presence a lot better.

PaulH

- Original Message - 
From: Trevor Peirce [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 12, 2005 8:57 AM
Subject: [Asterisk-Users] Presence Fully Supported?

 I've seen lots about presence and Polycom phones recently. I've got  one
 here for evaluation but noticed other phones only seem to appear busy
 when they initiate a call. If they receive a call, they still show as
 available.

 Is this a config problem on my part, or is that as far as presence is
 working right now?

 Thanks!
 Trev
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[Asterisk-Users] first character in line 11 missing

2005-09-11 Thread Ronald Wiplinger


I would like to know if somebody else experienced that:

sip show peers will always drop the first character of the 11th line.

while   sip show peers like [0-9,a-z]  will not drop any character.


Can anybody test this, please?


bye

Ronald Wiplinger

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Re: [Asterisk-Users] Registered SIP '202' ... expires 1800. Why does it expire

2005-09-11 Thread Tony Hoyle


Olle E. Johansson wrote:


If you are not roaming, set host=ipaddress of the phone and disable
registration in the phone. Then Asterisk will always know where the
phone is.

Not roaming, but it is DHCP based and fixing it would be problematic due 
to the way the network is setup.


I'll just continue regularly rebooting the phone for now...

Tony
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Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

2005-09-11 Thread C. Hatton Humphrey

 I think the biggest thing that hurts ham radio's ability to react to a
 crisis is the lack of equipment and operators.  Most of the traffic we pass
 is Health and Welfare with Logistics being the second to it. 

You might be interested to take a listen to the latest ARRL News -
they give a count of Priority traffic messages passed for Katrina...

http://www.arrl.org/arrlletter/audio/

The site is ARRL and it's their ARRL Letter feed to be presented on
repeaters.  The ARES response to Katrina articles have the info I'm
referring to.

Sorry for the OT addition to the thread but I find it worth
mentioning.  Also, for my two cents I'll toss in that the first thing
I thought of when someone mentioned using Asterisk with Ham was to get
a Laptop with a WiFi connection, Asterisk and a radio interface on
scene to provide comm links.

73 de NY5I
Hatton Humphrey
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RE: [Asterisk-Users] TE406p no interrupts

I modified wct4xxp.c and installed it.
This is the message for 'modprobe wct4xxp'

--
FATAL: Error inserting wct4xxp
(/lib/modules/2.6.9-1.667smp/extra/wct4xxp.ko): No
such device
FATAL: Error running install command for wct4xxp
astpbx kernel: Oops:  [1] SMP 
astpbx kernel: CR2: a0362081

Regards,
Jason

--- Boris Bakchiev [EMAIL PROTECTED] wrote:

 You should have just done this:
 rmmod wct4xxp
 rmmod zaptel
 modprobe wct4xxp
 
 It will do the same thing
 
  -Original Message-
  From: [EMAIL PROTECTED]
 [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Jason Kim
  Sent: Monday, 12 September 2005 00:34
  To: asterisk-users@lists.digium.com
  Subject: RE: [Asterisk-Users] TE406p no interrupts
  
  I modified wct4xxp.c and make clean; make linux26;
  make install; reboot;
  But the system is not rebooted.
  Because the system is in remote office I will
 check it
  next morning.
  Could you let me know your linux version, *
 version
  and motherboard?
  
  Thank you Boris.
  
  --- Boris Bakchiev [EMAIL PROTECTED] wrote:
  
   Well.
   Try this please (but only if you're running on
 the
   latest sources).
   Open wct4xxp.c sources and search for
   pci_module_init
   Replace it with pci_register_driver
   So the line should read:
   res = pci_register_driver(t4_driver);
  
   That allows you to get the card working on
 2.6.13 in
   almost exactly the
   same setup as yours.
  
   One weird thing though. Do no use insmod
   ./wct4xxp.ko from zaptel
   directory as it will not work. Do a proper make
   install and then
   modprobe.
  
  
   This is just part of the fixes you might need to
 do.
   If you encounter a problem after span
   reconfiguration (ztcfg) let me
   know.
  
   If you get stuck.. let me know.
  
   Regards
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED]
 On
   Behalf Of Jason Kim
   Sent: Sunday, September 11, 2005 8:14 PM
   To: asterisk-users@lists.digium.com
   Subject: RE: [Asterisk-Users] TE406p no
 interrupts
  
   I'm using FC3.
  
   uname -a
   -
   Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2
   15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux
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RE: [Asterisk-Users] Registered SIP '202' ... expires 1800. Why doesit expire

2005-09-11 Thread Race Vanderdecken

Because the server does not want dead or unconnected phones that might
move.

If the phone does not send a REGISTER every so often, periodically, then
the server will assume the phone is no longer available to send calls
to.

Unlike the Government, Asterisk will not send checks to dead people.

Race the tyrant Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: Friday, September 09, 2005 10:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Registered SIP '202' ... expires 1800. Why
doesit expire

Hi,

When a SIP client registers on Asterisk server, why it expires after
certain amount of time?

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RE: [Asterisk-Users] TE406p no interrupts

Well.
That means pci_register_driver probably not ding what it supposed to do.
In newer kernels pci_module_init should be replaced with
pci_register_driver as pci_module_init doesn't it what it supposed to.
How brave are you at getting a new kernel on your system?
I'm currently running on 2.6.13 on 955X chipset and it works really
well.
At first I had all sorts of problems with interrupts but with couple of
patches to wct4xxp all working just fine with close to 3-5K of calls per
day.

What is the model of the motherboard you have?
See if you can force a particular IRQ on a slot where your TE406P is.
Some motherboards do allow this, so you can assign IRQ bellow 15 to the
card.
That could help as well.
For now, revert the changes back. If you can, try new kernel (in
parallel) with the pci_register_driver.
Regards


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jason Kim
 Sent: Monday, 12 September 2005 11:28
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] TE406p no interrupts
 
 I modified wct4xxp.c and installed it.
 This is the message for 'modprobe wct4xxp'
 
 --
 FATAL: Error inserting wct4xxp
 (/lib/modules/2.6.9-1.667smp/extra/wct4xxp.ko): No
 such device
 FATAL: Error running install command for wct4xxp
 astpbx kernel: Oops:  [1] SMP
 astpbx kernel: CR2: a0362081
 
 Regards,
 Jason
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Re: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups

2005-09-11 Thread Matt Riddell

Chee Foong wrote:
 i guess may be it's a 64bit variable. so you can only use 0-63.

LOL

Bits work like this

[128][64][32][16][8][4][2][1]

So, you have a whole lot of bits, each one moving from right to left inceases
a power of 2.  Say you wanted to represent 10, then you would turn on the 8
and the 2 (8+2=10) so the binary representation (for an 8 bit variable) would 
be:

1010

So, from that you can see that you could get from 0-255 in 8 bits.

If however you wanted the number to be able to go negative as well then you
would use one of the bits to determine the sign (i.e. +/-)

That would give you possible values between -127 and +127.

So, a max value of 63 would either indicate a signed 7 bit variable (dunno
where the other one went) or an unsigned 6 bit variable.

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups

2005-09-11 Thread Paul


Matt Riddell wrote:


Chee Foong wrote:
 


i guess may be it's a 64bit variable. so you can only use 0-63.
   



LOL

Bits work like this

[128][64][32][16][8][4][2][1]

So, you have a whole lot of bits, each one moving from right to left inceases
a power of 2.  Say you wanted to represent 10, then you would turn on the 8
and the 2 (8+2=10) so the binary representation (for an 8 bit variable) would 
be:

1010

So, from that you can see that you could get from 0-255 in 8 bits.

If however you wanted the number to be able to go negative as well then you
would use one of the bits to determine the sign (i.e. +/-)

That would give you possible values between -127 and +127.

So, a max value of 63 would either indicate a signed 7 bit variable (dunno
where the other one went) or an unsigned 6 bit variable.

 


But I am on the other side of the equator from you!

Should I move from left to right or should I reverse the polarity to get 
the same results?


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[Asterisk-Users] Asterisk on AMD64

2005-09-11 Thread Joseph

Does anybody runs Asterisk on AMD64?

I can compile it on Gentoo, and start Asterisk a command line but as
soon as I connect any device (like Sipura ATA ), asterisk crashes.

-- 
#Joseph
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[Asterisk-Users] Syslog file size

2005-09-11 Thread YT Lim

Does anyone know how to limit syslog file size?
Logrotate only rotates log files (i.e. irrelevant of
file size), and a log file size can grow extremely
large before it is rotated.

/Y.T.






 
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RE: [Asterisk-Users] Syslog file size

2005-09-11 Thread Brad Hughes

Not possible to do natively

Write a script yourself to monior it's size and invoke logrotate to
rotate it when that size is reached. And cron your script however often
you want (ie 10 times a day?)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of YT Lim
Sent: Monday, 12 September 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Syslog file size


Does anyone know how to limit syslog file size?
Logrotate only rotates log files (i.e. irrelevant of
file size), and a log file size can grow extremely
large before it is rotated.

/Y.T.






 
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The New Yahoo! Movies: Check out the Latest Trailers, Premiere Photos
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RE: [Asterisk-Users] Syslog file size

2005-09-11 Thread YT Lim

Thanks. I'm afraid that's the case. But I remember
reading somewhere that some syslog variants of syslogd
actually have the file size limit build-in.


--- Brad Hughes [EMAIL PROTECTED] wrote:

 Not possible to do natively
 
 Write a script yourself to monior it's size and
 invoke logrotate to
 rotate it when that size is reached. And cron your
 script however often
 you want (ie 10 times a day?)
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of YT Lim
 Sent: Monday, 12 September 2005 1:01 PM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: [Asterisk-Users] Syslog file size
 
 
 Does anyone know how to limit syslog file size?
 Logrotate only rotates log files (i.e. irrelevant of
 file size), and a log file size can grow extremely
 large before it is rotated.
 
 /Y.T.
 
 
   
 
   
   
 
 
 Do you Yahoo!? 
 The New Yahoo! Movies: Check out the Latest
 Trailers, Premiere Photos
 and full Actor Database. 
 http://au.movies.yahoo.com
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[Asterisk-Users] extensions.conf for VOXEE using SIP!!

2005-09-11 Thread Jerry James


Hello,

I have been trying to setup a Voxee Sip termination. If anyone has 
extensions.conf different than Voxee suggestion.

Can you please send me a copy?

Thanks!

Jerry

Voxee  web site advises to use:

[voxee]
exten = _1NXXNXX,1,Dial,SIP/${EXTEN}voxee
exten = _1NXXNXX,2,Hangup
exten = _011.,1,Dial,SIP/${EXTEN}voxee
exten = _011.,2,Hangup

register=userid:[EMAIL PROTECTED]

sip.conf Settings

[voxee]
type=friend
username=userid
secret=password
host=66.246.246.52
fromuser=userid
dtmfmode=rfc2833
*

What I receive when dialing is:
- Executing Dial(SIP/5000-d1a7, SIP/19165551212) in new stack
Sep 11 23:32:04 WARNING[16363]: chan_sip.c:1398 create_addr: No such 
host: 19165551212

Destroying call '[EMAIL PROTECTED]'
Sep 11 23:32:04 NOTICE[16363]: app_dial.c:759 dial_exec: Unable to 
create channel of type 'SIP'


I have tries several modifications including changing first line to:

exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])

This will shut down asterisk on the first call!!

Can anybody share with me their  extensions.conf?

Thanks,

Jerry


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RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM




I can understand that. I'm a KL7 call so comms could 
mean the matter of someone getting picked up or freezing to 
death.

It troubles me that radio site owners (the ones who hold 
the pink slip on the tower and hilltop) are not providing power. In AK, 
most of these sites are multihomed
with fed, state and local radio systems so money is 
provided to maintain backup power.

That being said, in that given area, maybe taking a cue 
from the Emergency Call boxes along the I-5 and I-15 and use solar panels to 
charge a battery backup system. That plus some power-stingy equipment 
could maintain a reliable radio network. Knowing that all of us on the 
west coast are just || close to the big one when sites like this loose power to 
the cellular equipment, guess who's still going to be operating? :) (not 
that they would be working well anyways since lines jam up)

Anyways. A resiliant recovery plan that has been 
practiced and works will trump a "all-hands" effort anyday.

-Don



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D 
SchelinSent: Sunday, September 11, 2005 2:46 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] civil emergency comms: Asterisk + HAM
Don, I agree with you on many fronts. I come from a radio background 
and here in southern cal unless we fall into the sea nothing will take out all 
of the communications here including ham because we are not in low lying flat 
land and were too diversified, over 150 miles and as many mountain top sites. 
BUT,let me tell you about how bad the southern CA. radio site owners are 
becoming. We had a 4 day outage at a very large site where one of my radios is 
located. None of them care anymore about backup power. This happened this past 
week. We took up our own Generator because the site owner (a national site 
company) won't maintain an old one. My friend (a microwave isp ) fixed the 
site owners by adding oil and a new battery. That will take us 
out!Don Fanning wrote:

  
  Time and time again, emergency action drills take place 
  in cities to target where their weaknesses are in "crisis" handling. 
  Usually they involve planes crashing or explosions (mock of course). 
  Obviously they were never prepared for this sort of disaster in their recovery 
  plan. I've participated in a few ARES/RACES drills and have to say that 
  much could be done to improve upon the "HAM" 
  infrastructure.
  
  Most of the time, communications is coordinated through 1 
  repeater system. When this repeater goes down, of course people would 
  switch comms to another but in a case like this, where all the repeater 
  systems go down except for maybe one, there needs to be a better 
  plan.
  
  In Amateur Satellite Service, these orbiting "Repeaters" 
  employ a system called RUDAK where a chunk of spectrum is repeated. 
  Obviously this isn't feasible in terrestrial repeaters but they dohave 
  the ability to turn off radios and switch bands at will depending on operating 
  conditions. With software controlled radio and Asterisk, the repeater 
  system could be made to be more resilient to disaster by linking to other 
  repeater systems via radio where it could connect outward. 
  
  
  If you figure the overhead of a repeater's transmitter 
  and receiver plus the controller, replaceing the controller with an asterisk 
  based unit (integration) would make more sense as it would give the repeater 
  system much more capabilities in the same footprint and power. 
  Additionally, these repeater systems are located on hilltops with other radio 
  systems so they should have emergency power available (if you've ever been to 
  a hilltop repeater site, you'll know what I mean). 
  
  I think the biggest thing that hurts ham radio's ability 
  to react to a crisis is the lack of equipment and operators. Most of the 
  traffic we pass is "Health and Welfare" with "Logistics" being the second to 
  it. What defeats this is that in a disaster where local/high band long 
  haul capabilities are diminished, is simply the one repeater that is 
  functional because everything is squeezed onto one VHF/UHF 
  repeater.
  
  Where I could see thing being improved? 
  Installation of 802.11b/g WLAN under Part 97. It would allow for more 
  users into the system, there are less hardware and power components and allows 
  the system to be dynamically configured. Asterisk could play a huge role 
  then as it's made for IP based traffic and could re-route in a split 
  second.
  
  -Don
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of Michael D SchelinSent: Saturday, September 10, 
  2005 10:20 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] civil emergency comms: 
  Asterisk + HAMThe two best forms of communications in a 
  real disaster and one always has been is #1 Ham radio. and #2 satellite 
  telephone. Ham radio is global and has

RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM




Try contacting the repeater trustee for http://www.wa3key.com/blura.html. 
They have a picture of one on their site with it lit up.
You will need to recrystal the radio to a proper TX/RX 
pair for 70cm. However, depending on your area, you should contact your 
local repeater coordnator so you don't step on anyone's toes (especially the 
case in So.Cal ;)

Looks like you can order crystals from: http://www.icmfg.com/motorola.html.

And there are plenty of links associated with this 
hardware. Google is your friend.

As for interfacing it to *, you'll need a phone patch 
adapter. You could purchase one or build one but you'll need to get more 
information on how to do such.
Once you have the repeater up and running, you also 
need to setup * to see the phone patch/radio interface as a radio. This 
may require a controller card. (see the voip-info.org wiki) And... 
if you're going to go that far, consider enrolling into the echoirlp 
project. It's a VoIP oriented repeater link system that uses the internet 
as it's conduit. By Part 97 rule, the system must be protected from 
unlicensed use so interfacing with asterisk would require password protection 
and you as the repeater owner would be liable for any misuse of the 
system.

73 de Don 



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Steve 
TotaroSent: Sunday, September 11, 2005 6:07 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] civil emergency comms: Asterisk + HAM

Just a shot in the dark here. 

I bought this unit http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5792377951rd=1sspagename=STRK%3AMEWN%3AITrd=1a 
couple months ago hoping to connect it to an * system for experimentation. 
I am a HAM n00b. I can found no documentation on this unit anywhere. 
Does anyone know where to start?

I joined a local HAM club but have not had any time 
to go and pick brains. I am afraid to really even plug it in until I know 
what I am doing and have a call sign and everything so the FCC does't kick in my 
door. I did plug it in for a minute and there were no lights or anything 
so I not even sure it works.

Anyone have any links or ideas?

Thanks,
Steve

  - Original Message - 
  From: 
  Don Fanning 

  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Sunday, September 11, 2005 1:37 
  PM
  Subject: RE: [Asterisk-Users] civil 
  emergency comms: Asterisk + HAM
  
  Time and time again, emergency action drills take place 
  in cities to target where their weaknesses are in "crisis" handling. 
  Usually they involve planes crashing or explosions (mock of course). 
  Obviously they were never prepared for this sort of disaster in their recovery 
  plan. I've participated in a few ARES/RACES drills and have to say that 
  much could be done to improve upon the "HAM" 
  infrastructure.
  
  Most of the time, communications is coordinated through 1 
  repeater system. When this repeater goes down, of course people would 
  switch comms to another but in a case like this, where all the repeater 
  systems go down except for maybe one, there needs to be a better 
  plan.
  
  In Amateur Satellite Service, these orbiting "Repeaters" 
  employ a system called RUDAK where a chunk of spectrum is repeated. 
  Obviously this isn't feasible in terrestrial repeaters but they dohave 
  the ability to turn off radios and switch bands at will depending on operating 
  conditions. With software controlled radio and Asterisk, the repeater 
  system could be made to be more resilient to disaster by linking to other 
  repeater systems via radio where it could connect outward. 
  
  
  If you figure the overhead of a repeater's transmitter 
  and receiver plus the controller, replaceing the controller with an asterisk 
  based unit (integration) would make more sense as it would give the repeater 
  system much more capabilities in the same footprint and power. 
  Additionally, these repeater systems are located on hilltops with other radio 
  systems so they should have emergency power available (if you've ever been to 
  a hilltop repeater site, you'll know what I mean). 
  
  I think the biggest thing that hurts ham radio's ability 
  to react to a crisis is the lack of equipment and operators. Most of the 
  traffic we pass is "Health and Welfare" with "Logistics" being the second to 
  it. What defeats this is that in a disaster where local/high band long 
  haul capabilities are diminished, is simply the one repeater that is 
  functional because everything is squeezed onto one VHF/UHF 
  repeater.
  
  Where I could see thing being improved? 
  Installation of 802.11b/g WLAN under Part 97. It would allow for more 
  users into the system, there are less hardware and power components and allows 
  the system to be dynamically configured. Asterisk could play a huge role 
  then as it's made for IP based traffic and could re-route in a split 
  second.
  
  -Don
  
  
  
  From:

[Asterisk-Users] Anyone using Telasip, Caller ID presentation outbound??

2005-09-11 Thread Chris Coulthurst




II noticed that Caller ID presentation is not 
making it to my cell phone through outound Telasip calls and I don't know 
why. It may very well have been this way for awhile (or always, not sure I 
called my cell phone during telasip testing). Does Telasip expect a 
different format than SetCIDNum(NXXNXX) ? It hasalways worked 
for the Teliax lines.

BUT---
 It doesn't have a problem making 
it to landline phones Ive tried...

I user Verizon for the cell and Qwest for my 
incoming analog (with callerID) lines...

Chris Coulthurst
[EMAIL PROTECTED]

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RE: [Asterisk-Users] civil emergency comms: Asterisk + HAM

Priority traffic by ARRL standards would fall into both of these
categories.  What they are saying is that if someone is in a area where
a ham is operating and needs to get someone hauled out via emergency
services, priority traffic would take precedence over normal traffic.
Not quite a Mayday situation but close to.   Hams have come through
for the most part but since we're way off topic, it boils down to poor
planning on the emergency coordinator for a given
town/county/city/state.  

Let's face it.  When FEMA rolls in, there's no question about their
communications.  If they can run it through commercial terrestrial
providers, fine.  Otherwise, they have satellites phones that take less
than a few minutes to set up (if that).  Sure it's expensive to joe
smith.  But we're talking about the government here where justification
always outweighs cost.

That being said.  Asterisk has tremendous value to the HAM community.
People have always been happy to get a phone call from a serviceman at
sea (using MARS) or using autopatches to order pizza's.  I don't think
that part is argued.  The question is how it could be helpful?

Asterisk Conferences - Add the ability for people who are HAMS to log
into a protected chat room and communicate to both equipped and non
equipped hams (using cell phones).  Emergency services could
teleconference a Public Radio Service repeater and monitor the
conference to coordinate responses with lower overhead (again using COTS
equipment).

Asterisk Autopatching - This would allow people to setup Health and
Welfare phone booths for people to call their loves ones and coordinate
their return to a normal life.  One feature that I see really lacking in
Asterisk however is the ability to outdial from a teleconference to
three-way them into a conference as well as moderator functions.  Of
course these features are in Alliance teleconferences but would be nice
to add in as well.

Cepstral Integration - Imagine if your car was stolen and it was
equipped with APRS.  You could write a script that would read lon/lat,
do the map lookup and feed back location information every 10 seconds to
assist in recovery.  All it would take is 3-waying into the asterisk,
logging in and having * read back the information to emergency response.

The applications are endless with a system like this.

-Don
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Hatton
Humphrey
Sent: Sunday, September 11, 2005 6:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM

 I think the biggest thing that hurts ham radio's ability to react to a

 crisis is the lack of equipment and operators.  Most of the traffic we

 pass is Health and Welfare with Logistics being the second to it.

You might be interested to take a listen to the latest ARRL News - they
give a count of Priority traffic messages passed for Katrina...

http://www.arrl.org/arrlletter/audio/

The site is ARRL and it's their ARRL Letter feed to be presented on
repeaters.  The ARES response to Katrina articles have the info I'm
referring to.

Sorry for the OT addition to the thread but I find it worth mentioning.
Also, for my two cents I'll toss in that the first thing I thought of
when someone mentioned using Asterisk with Ham was to get a Laptop with
a WiFi connection, Asterisk and a radio interface on scene to provide
comm links.

73 de NY5I
Hatton Humphrey
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[Asterisk-Users] Asterisk and AMP installed now what?

2005-09-11 Thread Tommy Denton

Ladies and Gentleman,

I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going.

I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP.

Is there some docs somewhere on how to confgure once I have AMP up and running? 

I am not a telephony guy and alot of this looks like greek to me.
I have about 4 hours of tinkering under my belt and I am at the point I
need some help.

Thank you for your time,

Tommy
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[Asterisk-Users] Calling from one port on a SIPURA 2002 to the other port.

2005-09-11 Thread Paul Conn

I've been burning the midnight oil trying to configure Asterisk for the
first time.  If you have a 2 port SIPURA 2002 can you call from line 1 back
to line two?  I have a standard two line, DTMF telephone connected to both
ports.  Both lines one and two ARE registered and I can get and leave VM on
either one but I cannot dial the line 2 extension (6101) from line one
(6100).

Thanks!

Paul

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Re: [Asterisk-Users] Asterisk and AMP installed now what?

2005-09-11 Thread Chris




 I don't use Voiceplus so I don't 
have the settings. Did you look in the Wiki?


Chris


  - Original Message - 
  From: 
  Tommy 
  Denton 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Sunday, September 11, 2005 11:50 
  PM
  Subject: [Asterisk-Users] Asterisk and 
  AMP installed now what?
  Ladies and Gentleman,I have setup Asterisk and 
  AMP. They are working with out error. But now I need to get 
  everything going.I have Voicepluse and they give sample iax.conf and 
  extensions.conf files but that does me a little good as I am using 
  AMP.Is there some docs somewhere on how to confgure once I have AMP up 
  and running? I am not a telephony guy and alot of this looks 
  like greek to me. I have about 4 hours of tinkering under my belt and I 
  am at the point I need some help.Thank you for your 
  time,Tommy
  
  

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Re: [Asterisk-Users] Syslog file size

2005-09-11 Thread Samy Antoun

Hi,

This page
http://linuxcommand.org/man_pages/logrotate8.html

has a sample config file with a file size option






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RE: [Asterisk-Users] TE406p no interrupts