date:20051118

2005-11-18 Thread Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas


Hi

i would like to create my own soft Phone
for my local office use

can any one guide me the URL for the same
of source Soft Phone
or resources to create

ram
___
--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

Re: [Asterisk-Users] List of Motherboards or Servers that are testedok with Asterisk and Digium boards

2005-11-18 Thread Clive

Atari 600XL.16K ram...lol



On 18 Nov 2005 at 19:00, [EMAIL PROTECTED] wrote:

 I went from a Vic20 to a CPC6128...both great items
 
 PaulH
 
 - Original Message - 
 From: Julian Lyndon-Smith [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Friday, November 18, 2005 6:49 PM
 Subject: Re: [Asterisk-Users] List of Motherboards or Servers that are
 testedok with Asterisk and Digium boards
 
 
  Man, looking back it was a gas - the 16k wobbly rampack. You spent 30
  minutes looking at a blank screen whilst loading Horace goes skiing
  (or some other c*appy game you wanted to hack) making incantations to
  the tapedrive god in the vain hope that you wouldn't get an I/O error.
 
  It wasn't a gas then. ;)
 
  Always coveted a 48k Spectrum. Got a CPC6128 instead ..
 
  Julian.
 
  Matt Riddell wrote:
   Julian Lyndon-Smith wrote:
   Asterisk is cool. But maybe not that cool.
  
   Hey, don't you know that the dev team gets all the cool toys ;)
  
   You can tell I started coding on a ZX81.
  
   Woohoo go the ZX81!!!
  
 
  ___
  --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 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 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

Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service

2005-11-18 Thread Frederic Steinfels


Florian Overkamp wrote:


Hi Frederic,

Not to start some flame war here, but I've always known the Junghanns 
people to be quite cooperative, although it is a shame that they don't 
have two Klaus'es around there, since one is just simply too busy :)


Klaus was always very very kind on the phone. The only thing is that he 
almost never returned my calls and that I had to try 20 times to get 
hold of him once. On the other hand his brother just ditched me and did 
not even want to talk back with Klaus. If we make a mistake in our 
company, we have to apologize and try to agree on a compromise. That 
idea never occured to Jens. And that's what is upsetting me most because 
fighting for the return, buying another product, spending all the time 
on the installation etc. is barely worth it.


Frederic
___
--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

Re: [Asterisk-Users] Re: Can anyone explain reason for this echo

2005-11-18 Thread Florian Overkamp


Hi Eric,

Eric Bishop wrote:

I purchased the following item:
 http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html

As you can see not a very highly spec'd product but does the job well.


Can you indicate price range for this unit ?

Florian
___
--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

Re: [Asterisk-Users] Anyone know who is in this picture?

2005-11-18 Thread pdhales

I couldn't find his bio on rotten.com

- Original Message - 
From: Greg Boehnlein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 18, 2005 5:05 AM
Subject: Re: [Asterisk-Users] Anyone know who is in this picture?


 On Wed, 2 Nov 2005, Matt Darnell wrote:

  Well that didn't take long!
 
  He was a really nice guyI bet it would be a blast to go have a beer
with
  him.
 
  We met him at the Internet Telephony Expo.

 Read his bio on Rotten.Com. I'm surprised to see him posing with Women.

 -- 
 Vice President of N2Net, a New Age Consulting Service, Inc. Company
  http://www.n2net.net Where everything clicks into place!
  KP-216-121-ST



 ___
 --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 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

Re: [Asterisk-Users] create my own soft Phone

2005-11-18 Thread Damian Minkov


ram wrote:



Hi
 
i would like to create my own soft Phone

for my local office use
 
can any one guide me the URL for the same

of source Soft Phone
or resources to create
 
ram



http://sip-communicator.org/
___
--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

Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service

2005-11-18 Thread Frederic Steinfels


Matt Riddell wrote:


Not if he was told to wait for it to work.


thanks. that was the case.
you see my warrranty claims were made within the first weeks after 
getting the product. I was patient for two years. I am still patient 
because fighting for 600euros is barely worth it. so I still appreciated 
it if they released a version that was working without destroying the 
isdn connection every now and then (at least bristuff is no longer core 
dumping asterisk as it did frequently for over a year).


furthermore there is a two year warranty by law in germany.
___
--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

Re: [Asterisk-Users] Hung Zap channels




Hmm, it sounds to me the hangup detection problem. 
Here's a way to solve that problem:
Applythis patch:http://www.maxosystem.net/asterisk/asterisk-stable-polarity-v5.diff
$ cd /usr/src/asterisk/channels$ patch 
chan_zap.c  /your/route/here/asterisk-stable-polarity-v5.diff

and in zapata.conf :

answeronpolarityswitch=yeshanguponpolarityswitch=yes


Sure, it will only help if your PSTN company does hangup detection by 
changing polarity of the line, refer to their line specificacions to more 
info.

Hope it helps ;)


- Original Message - 
From: "John Heng" [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 18, 2005 1:56 
AM
Subject: [Asterisk-Users] Hung Zap 
channels
Hi all,I'm running asterisk 1.0.9 (yes I know - 1.2 has just been 
released) with a TDM400P board that has 4 FXO port. Once in a while, I've found 
that the zap channel will get stuck (or blocked) even after the call has ended. 
Sometimes this is when someone has left a voice msg, but not always. The 
way I've fix this is to issue a "soft hangup" command for that zap channel. 
However, I'm not always aware of this until a user tells (or complains to) me. 
What I would like to know is if there is a way to reset all the zap 
channels or re-initialize the drivers without restarting Asterisk. If so, I 
could set up a cron job to do it once or twice a week, in the middle of the 
night. Any suggestion guys??CheersJ 
Heng___--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
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] create my own soft Phone

Hi

i would like to create my own soft Phone
for my local office use

can any one guide me the URL for the same
of source Soft Phone
or resources to create

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] re: problem with asterisk and SIP on same box with 1.2

2005-11-18 Thread Yair Hakak

hello all,
 having a little problem..
 asterisk and ser on the same box, SER on 5060 and asterisk on 5070.
SER is set up to forward everything to asterisk.

in 1.07 my sip.conf looked like this:

[general]
port = 5070 ; Port to bind to
disallow=all; Disallow all codecs
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
dtmfmode=rfc2833

context=myUsers
canreinvite=no
host=dynamic
insecure=no
nat=yes
qualify=1000

autocreatepeer=yes

and incoming SIP requests flowed to asterisk.

now, it's failing, silently, with nothing in the CLI (at v).
ngrep the sip packets show SER trying to forward the packets along and failing.

anyone have something similar or have any tips? do i need to add insecure?

thanks,
 yair
___
--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

Re: [Asterisk-Users] ip phone

2005-11-18 Thread trixter aka Bret McDanel

On Fri, 2005-11-18 at 15:04 +0700, stevanus wrote:
 Hi,
 
 Maybe grandstream budgetone 100 series will fulfill your requirement.
 It's very good for such a cheap sub-50 phone.
 Once, I've tested and I've found myself that it's a good performer
 (even it has compatibility problem with old switch in my office :P)
 You can search the supplier through googling it. Don't ask me as I
 don't know any information about it.

I have heard bad things about that phone.  Specifically audio quality is
questionable, the power connector that ships is the wrong size so it
tends to fall out, there are firmware issues that locks the phone up,
etc.  

Does anyone have any experience with that phone specifically?

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] gpx-2000 early dial support

2005-11-18 Thread Louis-David Mitterrand

Hi, 

The gxp-2000's lack of a dialplan (or did I miss it?) led me to activate
its early dial option to avoid pressing Send after dialing. Thus the
dialplan is controlled by asterisk.

It creates an extension matching problem:

exten = _00[1-9].,1,Macro(dialcapi)

If I dial 0012 the extension is matched immediately. Is there a way to
ask asterisk to wait a few seconds for more digits?

Thanks,
___
--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

Re: [Asterisk-Users] create my own soft Phone

2005/11/18, ram [EMAIL PROTECTED]:
 can any one guide me the URL for the same
 of source Soft Phone
 or resources to create

Look in freshmeat.net for some soft phones, and download the source codes.

But... if you want to work programming a soft phone, why duplicate
efforts. Choose the project nearest your needs and help the team to
improve it to meet what you need. It always will be better than
working alone without help.

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Newbie question. (Long)


Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
downloaded the binary package.


Now I'm trying to put the working installation on my production server 
along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried the 
tarball download of 1.2.0.rc2, ran make OK, then make install, make samples.

When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours later...
Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar 
ball again, and re-installed.

Same old problem, illegal instruction.

I did an strace, which follows. I don't know enough to decide what the 
strace is telling me. (The missing /etc/ld.so.preload is also missing on 
the FC4 laptop which works, so I concluded that that was not the problem.)


Any help much appreciated.

Regards
Roger

[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0)  = 0x8773000
access(/etc/ld.so.preload, R_OK)  = -1 ENOENT (No such file or 
directory)

open(/etc/ld.so.cache, O_RDONLY)  = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY)   = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000

close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, 
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000


close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000

close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000

close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY)   = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000

close(3)= 0
open(/lib/libssl.so.5, O_RDONLY)  = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x35000) = 0xadd000

close(3)= 0
open(/lib/tls/i686/libc.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\230n\177..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=1431008, ...}) = 0
old_mmap(0x7e2000, 1129660, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x7e2000

mprotect(0x8ef000, 27836,

[Asterisk-Users] Re: Re: SIP - Loop detected (Matt Riddell) (Matt Riddell)

2005-11-18 Thread Trond G. Andersen

Trond Andersen wrote:
 Thank you, but I have tried that... Then the To is:

Can you do a NoOp(${ARG1}) and then show us the result?

-- 
Cheers,

Matt Riddell


Thank you for taking the time to help me out Matt !


-- Executing NoOp(SIP/trond-c7f0, ARG1=20170) in new stack
-- Executing Dial(SIP/trond-c7f0, SIP/20170|30|Cf) in new stack
-- Called 20170

With this dialplan I do not get the loop, of course but my endpoint
needs the entire SIP-URI, so I must change the dialplan to be:

Dial (SIP/[EMAIL PROTECTED],30,Cf)

Then I get:
-- Executing NoOp(SIP/trond-c7f0, ARG1=20170) in new stack
-- Executing Dial(SIP/trond-c7f0, SIP/[EMAIL PROTECTED]|30|Cf) in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 482 Loop detected back from 10.47.8.89


10.47.8.89 is my asterisk IP and the domain name I want to use for my
extensions.


(I actually use ARG2 because ARG1 holds the extensions email addr. I
cannot imagine that matters??)


Thanks again,
Trond

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Missing smp kernel package in Asterisk 1.2 installation...

2005-11-18 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Leo Burd [EMAIL PROTECTED] wrote:
 Hello there,
 
 I've just downloaded Asterisk 1.2 into my RedHat Enterprise Linux 
 machine and got the following problem when I tried to compile zaptel:
 
 You do not appear to have the sources for the 2.6.9-22.ELsmp kernel 
 installed.
 
 However, according to rpm -qa, I do have the following packages 
 installed in my system:
 
 kernel-smp-2.6.9-22.EL
 kernel-smp-devel-2.6.9-5.EL

They don't match - you need kernel-smp-devel-2.6.9-22.EL

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problem switching from external ISDN-2 to PBX ISDN-2

2005-11-18 Thread Lars Dybdahl

I have a system that works perfectly using zaphfc on an external
ISDN-2 connection. When I move this to a PBX-based ISDN-2 connection,
it still receives calls, but is unable to dial out.

In Denmark, we have no long distance calls, but only national calls
(8-digit numbers) or international calls (00 countrycode etc.).
Normally, incoming national calls would be reported as 8-digit
numbers, but when I connect my asterisk to the PBX, it is preceded by
three zeroes (00012345678 instead of 12345678). This seems to indicate
a problem, too, since you normally only dial one zero on a PBX to get
an outside line.

Any ideas?

Lars.
___
--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

RE: [Asterisk-Users] ip phone

2005-11-18 Thread Lee Archer

We had 1 way speech on them for a while but the latest firmware seems to
have fixed it.  The 10mb LAN ports in the back are old too.  Also I
wouldn't recommend the GXP-2000 either.  We have a few here.  As a basic
phone it's fine but don't try anything fancy like PoE as ours keeps
failing and we have to run them off the PSU.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: 18 November 2005 09:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ip phone

On Fri, 2005-11-18 at 15:04 +0700, stevanus wrote:
 Hi,
 
 Maybe grandstream budgetone 100 series will fulfill your requirement.
 It's very good for such a cheap sub-50 phone.
 Once, I've tested and I've found myself that it's a good performer 
 (even it has compatibility problem with old switch in my office :P) 
 You can search the supplier through googling it. Don't ask me as I 
 don't know any information about it.

I have heard bad things about that phone.  Specifically audio quality is
questionable, the power connector that ships is the wrong size so it
tends to fall out, there are firmware issues that locks the phone up,
etc.  

Does anyone have any experience with that phone specifically?

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


###

This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.
For more information, connect to http://www.f-secure.com/
___
--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

Re: [Asterisk-Users] ip phone

2005-11-18 Thread Yair Hakak

look, you get what you pay for.
excellent value for the price, but i've found they need more
handholding than others (sometimes they need to be rebooted, they
freeze up, etc). i'm phasing out in favor of pap2 units and analog
phones.
i've never had a problem with audio quality, however, audio quality
with other devices is noticably better.

-yair

On 11/18/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
 On Fri, 2005-11-18 at 15:04 +0700, stevanus wrote:
  Hi,
 
  Maybe grandstream budgetone 100 series will fulfill your requirement.
  It's very good for such a cheap sub-50 phone.
  Once, I've tested and I've found myself that it's a good performer
  (even it has compatibility problem with old switch in my office :P)
  You can search the supplier through googling it. Don't ask me as I
  don't know any information about it.

 I have heard bad things about that phone.  Specifically audio quality is
 questionable, the power connector that ships is the wrong size so it
 tends to fall out, there are firmware issues that locks the phone up,
 etc.

 Does anyone have any experience with that phone specifically?

 --
 Trixter http://www.0xdecafbad.com Bret McDanel
 UK +44 870 340 4605   Germany +49 801 777 555 3402
 US +1 360 207 0479 or +1 516 687 5200
 FreeWorldDialup: 635378
 http://www.sacaug.org/ Sacramento Asterisk Users Group


 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.1 (GNU/Linux)

 iD8DBQBDfZnX+1olxlzQw5cRAio1AKCYSsEAlVOLrGCFTyHWNwJyMPDZLwCfeVKk
 B8cwwSFLC6Acs1eH4qV4Axg=
 =ITZY
 -END PGP SIGNATURE-


 ___
 --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 list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: how to originate a call and capture it's DIALSTATUS

2005-11-18 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Script Head [EMAIL PROTECTED] wrote:
 
 I've been trying to originate calls and capture the DIALSTAUS via the
 manager API. The problem seems that the API doesn't expose enough data to
 make a decision of what exactly happened to the call. It results in
 something like this:
 
 Action: Originate
 Channel: IAX2/switch/1number
 MaxRetries: 0
 WaitTime: 2
 Context: reminder
 Extension: s
 Priority: 1
 Callerid: Reminder 555-555-
 
 Event: Hangup
 Privilege: call,all
 Channel: IAX2/switch-3
 Uniqueid: 1132271784.42
 Cause: 0
 Cause-txt: Unknown
 
 this is far from detailed. Is there a way to extract the actual DIALSTATUS
 such as ANSWER,BUSY,CONGESION, etc? The Cause doesn't seem to return 0 when
 the call is terminted thru IAX2 or SIP. It seems that it works on ZAP only.

There are two things you could try.

1. Add Async: yes to the Originate action, and then watch for the
OriginateSuccess and OriginateFailure events.

2. If that doesn't reveal the wanted information, then I would consider
it a bug, but you could workaround it by using a Local channel:

[outgoing]
exten = _X.,1,Dial(IAX2/switch/1${EXTEN})
exten = _X.,2,UserEvent(Fail|Dialstatus: ${DIALSTATUS})

Then in your Originate action use Channel: Local/number@outgoing
and look for the UserEventFail event.

Hope this helps!
Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
--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

RE: [Asterisk-Users] HFC ISDN card and mISDN driver

2005-11-18 Thread Amatisoft SRL

I get the same errors with the install-misdn script from Beronet:Replying to myself to say that I solved this issue by downloading the CVS version of mISDN from isdn4linux's CVS repository and replacing the one from the tarball that gets downloaded by the Makefile.However, I still cannot get Asterisk to startup with mISDN, chan_misdn and an /etc/asterisk/misdn.conf file -- it keeps saying "init_stack: Function not implemented". If I remove the misdn.conf file, * will start, but won't initialise my card (obviously). I am successfuly using chan_misdn, the mISDN stuff from PBX4Linux and the 2.6.9 kernel to get multiple HFC-S cards in the same Asterisk box. You should try an older version of the Linux kernel.So, I'm back to CAPI and chan_capi-cm on * 1.0.9 until I can find an alternative ISDN-BRI card that allows multipl
e
instances in a single PC. Darn AVM!Fritz cards. *sigh*In order to get multiple instances of the AVM Fritz! PCI card in a single PC and use them with chan_capi and the AVM CAPI4Linux drivers, I had to patch the AVM's driver. I will try this days to write a small tutorial how to patch the 2.6 kernel and post it on the wiki.More details about the versions of the software I am using are here:http://amatisoft.homelinux.com/demo/cgi-bin/amatix/allpackages.html-- Amatisoft SRL http://amatisoft.homelinux.com
Yahoo! FareChase - Search multiple travel sites in one click.

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Streaming mp3's when dialing a particular extension.

2005-11-18 Thread Amith


hi all, 

i'm trying to Stream mp3's when dialing a particular
extension.
2000 in this case.

My last part of extensions.conf is as below :


snip

exten = 2000,1,Answer
exten = 2000,2,WaitMusicOnHold(30)
exten = 2000,5,Hangup

/snip

i'am able to reach exten = 2000,1,Answer. And i get a
200 Ok 
for the INVITE. But i see this error message on the
console.

res_musiconhold.c :309 monmp3thread: Request to
schedule in the
past?!?!

Could someone help me fix this ? 
Is this something to do with the timer ? 
I looked into res_musiconhold.c but couldn't
understand much of what was happening there ?

i wanted to stream a mp3 file , when the user dials a
certain extension , any clue on how to fix this or
better ways of doing it ? Any links would be great.


cheers,
Amith 








__ 
Yahoo! Mail - PC Magazine Editors' Choice 2005 
http://mail.yahoo.com
___
--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

RE: [Asterisk-Users] SER Asterisk combination to get around NAT

2005-11-18 Thread Stuart Hirst

Mark,

Thanks for your response.

The typical deployment is a single server in the customer location
directly on the end of an ADSL link with two Ethernet interfaces, 1 to
the ADSL modem and the other to the LAN. The LAN side is fine and is as
normal but many customers have remote users or remote small offices that
may have more than one SIP device behind NAT.

What I am trying to establish is how successful SER is at allowing
multiple remote SIP devices behind a remote NAT router to interact with
Asterisk and what issues need to be taken into account such as MWI and
or codec's.

I have been using Asterisk for quite some time but have not played with
SER yet and so does anyone have some sample SER configs to work in this
type of deployment.

Stuart


-Original Message-
From: Mark John Buenconsejo [mailto:[EMAIL PROTECTED] 
Sent: 18 November 2005 06:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SER  Asterisk combination to get around
NAT
Importance: High


Hello Stuart, we have, and I would be happy to help you setup both
Asterisk and SER on a consultancy basis.

You can find more information about me here:
http://mark.teamcebu.com

Basically, it requires SER to forward the SIP messages to Asterisk, and
that SER be configured as one of the SIP channels on Asterisk. What
happens is:

from the LAN Phone, it connects to SER 
and then SER forwards it to Asterisk 
Asterisk will connect to the actual destination 
As soon as Asterisk is able to connect to the destination, it then
replies to the phone that the call is connected 
At this point, the actual call connections are made (asterisk-to-phone
and asterisk-to-destination)

and then Asterisk bridges the asterisk-to-destination and
asterisk-to-phone connections 
The bridged call mechanism on Asterisk works around the NAT limitations 
In this setup, it will appear that the Phone is connecting to Asterisk
(LAN side), and that the destination is talking to Asterisk (Live side),
and Asterisk passes the RTP packets back-and-forth.

There are a few considerations though, such as codec supports. As much
as possible use the same codec for each leg of the call, otherwise the
call quality deteriorates during transcoding.

By the way, we're using this with up to 12 simultaneous calls in our
setup (a small call center), using either iLBC and G.729 codec.

Anyway, let me know if you need further help. :) Or if you have some
more specific questions.

Thanks!

Mark

Stuart Hirst wrote: 
Has anyone successfully used SER and Asterisk together on the same
server to get around NAT traversal issues.

I have looked at many of the NAT traversal topics which either involve
commercial products and significant costs or solutions such as STUN or
proprietary systems such as xten.

  



--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.3/174 - Release Date:
17/11/2005

-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.3/174 - Release Date:
17/11/2005
 

___
--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

Re: [Asterisk-Users] SIP - Loop detected

2005-11-18 Thread Olle E. Johansson

Trond,
You need to tell us more. The SIP phones - what are they registering as?
 (Show sip.conf peer configs)

If one register as a SIP peer trond you should be able to dial
SIP/trond and get a full URI. If not, something is really wrong.

/O
___
--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

Re: [Asterisk-Users] create my own soft Phone

Hi
thanks

i also mentioned in the mail
looking some resources to build also

suggest me some resources where to start over

ram
On 11/18/05, Alejandro Vargas [EMAIL PROTECTED] wrote:
2005/11/18, ram [EMAIL PROTECTED]: can any one guide me the URL for the same
 of source Soft Phone or resources to createLook in freshmeat.net for some soft phones, and download the source codes.But... if you want to work programming a soft phone, why duplicate
efforts. Choose the project nearest your needs and help the team toimprove it to meet what you need. It always will be better thanworking alone without help.--Alejandro Vargas___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users

___
--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

Re: [Asterisk-Users] Newbie question. (Long)

2005-11-18 Thread Vassil Kolarov


Hi Roger,

Following this instructions:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

I was able to install and run Asterisk several times without problems.

See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora

Regards,
Vassil Kolarov
www.ittconsult.com


Roger Hill wrote:

Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
downloaded the binary package.


Now I'm trying to put the working installation on my production server 
along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried the 
tarball download of 1.2.0.rc2, ran make OK, then make install, make 
samples.

When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours 
later...

Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar 
ball again, and re-installed.

Same old problem, illegal instruction.

I did an strace, which follows. I don't know enough to decide what the 
strace is telling me. (The missing /etc/ld.so.preload is also missing 
on the FC4 laptop which works, so I concluded that that was not the 
problem.)


Any help much appreciated.

Regards
Roger

[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0)  = 0x8773000
access(/etc/ld.so.preload, R_OK)  = -1 ENOENT (No such file or 
directory)

open(/etc/ld.so.cache, O_RDONLY)  = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY)   = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000

close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, 
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000


close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000

close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000

close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY)   = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000

close(3)= 0
open(/lib/libssl.so.5, O_RDONLY)  = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x35000) = 0xadd000

close(3)

RE: [Asterisk-Users] IAXmodem

2005-11-18 Thread Lee Archer

Title: IAXmodem



Thanks for the help. I got it working, 
type=friend

Lee



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee 
ArcherSent: 17 November 2005 13:53To: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] 
IAXmodem

Hi, I wonder if you can give me some pointers 
please. I have hylafax running, I've tested it with a modem off the serial 
port so I know the install does work, and I've installed IAXmodem to be able to 
fax out via asterisk. I've set everything up as in the README that comes 
with IAXmodem but im not getting the faxes sent. I can see hylafax sending 
to the IAXmodem but at this point something isn't working and I'm getting 

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: 
IAX Subclass: REJECT  Timestamp: 7ms SCall: 4 DCall: 29764 
[172.16.5.137:4569]  
CAUSE : No authority 
found  Unknown IE 042 : 
Present 
in the iax log and no dialtone in the hylafax 
log. 
The IAXmodem is setup in my asterisk as an IAX2 
extension. 
Any ideas? 
Regards 
Lee 
###This message has been 
scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, 
connect to http://www.f-secure.com/ 
___
--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

Re: [Asterisk-Users] PSGW 2.2 Skype gateway?

2005/11/17, Michael Graves [EMAIL PROTECTED]:
 Call quality is ok, but it seems to add considerable latency. I suspect
 that the call is fully decoded back to analogue (or maybe not quite
 that far) on one of the audio devices in the OS, then encoded into SIP
 for the outbound leg. That would imply additional delay in all cases.

It uses the skype api, then is the api (the skype propietary client)
who decodes the sound (adding some latency). Then the sip part may be
including some extra latency. It sould use a low latency codec,
compression is not needed in a local machine...

But there is a big problem with this program: it needs windows to run,
adding failure point to the circuit... and then needing of an extra
machine only for acting as gateway: bad solution.

I think we will need to wait until someone hacked the skype
protocol... If there is someone interested on doing it.


--
Alejandro Vargas
___
--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

Re: [Asterisk-Users] Problem switching from external ISDN-2 to PBX ISDN-2

2005-11-18 Thread amer karim

Hi;

What is ur extension.conf?
do u use overlapdial??
2005/11/18, Lars Dybdahl [EMAIL PROTECTED]:
I have a system that works perfectly using zaphfc on an externalISDN-2 connection. When I move this to a PBX-based ISDN-2 connection,
it still receives calls, but is unable to dial out.In Denmark, we have no long distance calls, but only national calls(8-digit numbers) or international calls (00 countrycode etc.).Normally, incoming national calls would be reported as 8-digit
numbers, but when I connect my asterisk to the PBX, it is preceded bythree zeroes (00012345678 instead of 12345678). This seems to indicatea problem, too, since you normally only dial one zero on a PBX to get
an outside line.Any ideas?Lars.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- 
cordialementKarim AMER 
___
--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

Re: [Asterisk-Users] Newbie question. (Long)


Thanks Vassil - I'll try those pointers and report back.

Roger

Vassil Kolarov wrote:


Hi Roger,

Following this instructions:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

I was able to install and run Asterisk several times without problems.

See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora

Regards,
Vassil Kolarov
www.ittconsult.com


Roger Hill wrote:


Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
downloaded the binary package.


Now I'm trying to put the working installation on my production 
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried the 
tarball download of 1.2.0.rc2, ran make OK, then make install, make 
samples.

When I tried to run Asterisk, I got (immediately) Illegal Instruction.
Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours 
later...

Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar 
ball again, and re-installed.

Same old problem, illegal instruction.

I did an strace, which follows. I don't know enough to decide what 
the strace is telling me. (The missing /etc/ld.so.preload is also 
missing on the FC4 laptop which works, so I concluded that that was 
not the problem.)


Any help much appreciated.

Regards
Roger

[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0)  = 0x8773000
access(/etc/ld.so.preload, R_OK)  = -1 ENOENT (No such file or 
directory)

open(/etc/ld.so.cache, O_RDONLY)  = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY)   = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000

close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, 
-1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000


close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000

close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000

close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY)   = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000

close(3)= 0
open(/lib/libssl.so.5, O_RDONLY)  = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE,

Re: [Asterisk-Users] multi tenant with queues

2005-11-18 Thread Lenz



You could use a prefix-based agent numbering scheme, like Agent/XXYYY  
where XX is your customer code and YYY their own agent number. When  
showing activity to a customer, you strip the XX part or you may leave it  
alone, as it makes no big confusion to the client.

Yours,
l.



On Fri, 18 Nov 2005 01:13:09 +0100, snacktime [EMAIL PROTECTED] wrote:


I'd like some feedback on my solution so far for using queues in a multi
tenant configuration. For most of the configuration files I've been able  
to

use a naming scheme for the context names, which works nicely for making
multi tenant fairly transparent. However that won't work for everything  
and

queues is one of them.

In queues.conf the naming scheme will work for defining a queue. It won't
work for the agents though as they all have to have unique names. My  
thought

is to create a pool of available agent numbers, and the web gui for the
tenants will let the tenant pick the agent numbers they want to assign  
out
of the pool. As numbers are used they are taken out of the pool, and as  
they
become available they go back into the pool. The downside to this is  
that a
tenant won't get to pick the exact numbers they want, but that doesn't  
seem

like too much of a compromise for a multi tenant system.

Anyone have any better ideas?

Chris




--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

___
--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

Re: [Asterisk-Users] PSGW 2.2 Skype gateway?

2005-11-18 Thread Francesco Peeters

On Fri, November 18, 2005 11:14, Alejandro Vargas said:
 2005/11/17, Michael Graves [EMAIL PROTECTED]:
 Call quality is ok, but it seems to add considerable latency. I suspect
 that the call is fully decoded back to analogue (or maybe not quite
 that far) on one of the audio devices in the OS, then encoded into SIP
 for the outbound leg. That would imply additional delay in all cases.

 It uses the skype api, then is the api (the skype propietary client)
 who decodes the sound (adding some latency). Then the sip part may be
 including some extra latency. It sould use a low latency codec,
 compression is not needed in a local machine...

 But there is a big problem with this program: it needs windows to run,
 adding failure point to the circuit... and then needing of an extra
 machine only for acting as gateway: bad solution.

 I think we will need to wait until someone hacked the skype
 protocol... If there is someone interested on doing it.



Which is going to be a pain, as it is encrypted...  :-(

Reverse-engineering may be the best option, and that is:
1) Not trivial
2) Not always legal

/me sighs...

I'll try getting my friends on to VoipBuster instead!  ;-)

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: chan_bluetooth

2005-11-18 Thread Victor Alvarez

Can you try this again with a CLI open on * with a high verbose level.
This is what I get when asterisk drops out of the chain.
 chan_bluetooth.c:701 sco_thread: SCO connection error: Connection
refused (errno 111)

This is the trace asterisk is giving me:


-
show version
Asterisk CVS-Nv1-0-9-11/14/05-15:26:22 built by [EMAIL PROTECTED] on a i686 
running
Linux
sip show peers
Name/usernameHostDyn Nat ACL Mask Port
Status
02/02(Unspecified)D  255.255.255.255  0
Unmonitored
01/01192.168.129.237  D  255.255.255.255  5060
Unmonitored
*CLI bluetooth show peers
BDAddrName   Role Status  A/C SCOCon/Fd/Th Sig
- --  --- ---  ---
00:0E:6D:7A:B1:FA N7600  AG   Ready   Yes -1/-1/0  Yes
00:12:62:E1:E5:45 Nokia  AG   Ready   Yes -1/-1/0  Yes
*CLI Nov 18 10:30:45 DEBUG[18269]: chan_sip.c:5456 check_user_full: Setting
NAT on RTP to 0
Nov 18 10:30:45 DEBUG[18269]: chan_sip.c:841 __c
'[EMAIL PROTECTED]' of Response 34150:
Found
Nov 18 10:30:45 DEBUG[18269]: chan_sip.c:5456 check_user_full: Setting NAT
on RTP to 0
Nov 18 10:30:45 DEBUG[18269]: chan_sip.c:7354 handle_request: Check for res
for 01
Nov 18 10:30:45 DEBUG[18269]: chan_sip.c:1623 update_user_counter: Call from
user '01' is 1 out of 0
Nov 18 10:30:45 DEBUG[18269]: chan_sip.c:4643 build_route: build_route:
Contact hop: sip:[EMAIL PROTECTED]:5060
-- Executing Dial(SIP/01-1724, BLT/N7600/XXX) in new stack
Nov 18 10:30:45 DEBUG[18312]: /usr/src/chan_bluetooth/chan_bluetooth.c:1112
blt_request: Dialing 'XXX' via 'N7600'
Nov 18 10:30:46 DEBUG[18312]: /usr/src/chan_bluetooth/chan_bluetooth.c:882
blt_call: Calling N7600 on BLT/N7600 [t: 0]
 [AG]  N7600  ATDXXX;
-- Called N7600
 [AG]  N7600  ATDXXX;
 [AG]  N7600  OK
Nov 18 10:30:47 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:1578
ag_unknown_response: Got UNKN response: OK
Nov 18 10:30:47 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 3 set to 2
 [AG]  N7600  +CIEV: 3,2
Nov 18 10:30:47 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 4 set to 2
 [AG]  N7600  +CIEV: 4,2
Nov 18 10:30:50 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 3 set to 3
 [AG]  N7600  +CIEV: 3,3
Nov 18 10:30:50 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 4 set to 3
 [AG]  N7600  +CIEV: 4,3
-- BLT/N7600 is ringing
Nov 18 10:30:52 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 1 set to 1
 [AG]  N7600  +CIEV: 1,1
-- BLT/N7600 answered SIP/01-1724
Nov 18 10:30:52 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 3 set to 0
 [AG]  N7600  +CIEV: 3,0
Nov 18 10:30:52 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 4 set to 0
 [AG]  N7600  +CIEV: 4,0
Nov 18 10:30:52 DEBUG[18269]: chan_sip.c:841 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of
Response 34151: Found
Nov 18 10:30:57 DEBUG[18312]: channel.c:2675 ast_channel_bridge: Didn't get
a frame from channel: SIP/01-1724
Nov 18 10:30:57 DEBUG[18312]: channel.c:2746 ast_channel_bridge: Bridge
stops bridging channels SIP/01-1724 and BLT/N7600
Nov 18 10:30:57 DEBUG[18312]: /usr/src/chan_bluetooth/chan_bluetooth.c:932
blt_hangup: blt_hangup(BLT/N7600)
 [AG]  N7600  AT+CHUP
Nov 18 10:30:57 DEBUG[18312]: app_dial.c:1054 dial_exec: Exiting with
DIALSTATUS=ANSWER.
  == Spawn extension (default, 04, 1) exited non-zero on 'SIP/01-1724'
Nov 18 10:30:57 DEBUG[18312]: chan_sip.c:1726 sip_hangup:
update_user_counter(01) - decrement inUse counter
 [AG]  N7600  AT+CHUP
 [AG]  N7600  OK
Nov 18 10:30:57 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:1578
ag_unknown_response: Got UNKN response: OK
Nov 18 10:30:57 DEBUG[18281]: /usr/src/chan_bluetooth/chan_bluetooth.c:386
set_cind: CIND 1 set to 0
 [AG]  N7600  +CIEV: 1,0

-
 ast_channel_bridge stops bridging channels SIP and BLT?

 Kind Regards,
  Victor.

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Contact field in SIP HF between asterisk + ser

2005-11-18 Thread harry gaillac

Hello,

Here is my config :


Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060 

I wish ser to handle the packets between Nat box
(netfilter) and  Asterisk However contact field  in
the sip HF don't change from nat box to asterisk which
don't allow to keep the sessions via SER .


Ser receive packets with private ip in contact field
which one is forward to asterisk .

How ser can handle the contact field to establish sip
sessions between sip agents and asterisk ? 

I've been trying mangle and textops modules but i
really need to be advice.


 

  One box
 ---
 |     |
 |  | asterisk pbx |   | 
 |     |
 |||   |
 |  ----
 |  |   SER  ||NAT box | private network
 |  ----
 ---

Regards
Harry






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
--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

Re: [Asterisk-Users] New asterisk management tool

2005-11-18 Thread Leif Neland


I need a hint:


From pbxmanager/doc/INSTALL


2.  Install a database adaptor via rubygems.  Postgresql, Mysql, and Sqlite3 
are all supported and tested to work.


Eh... How to install?

Leif


___
--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

Re: [Asterisk-Users] VoIP Gateway Providers

2005-11-18 Thread Nana Tandoh

try Ipkall.com or ipkall.netthey have DID for WA for free and also take a look at sellvoip.net or 
sellvoip.com. Let me know if you need any further help.
On 11/17/05, Kerry Garrison [EMAIL PROTECTED] wrote:
IAX.cc is what I use for my DID numbers.-Kerry-Original Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Jeff RamseySent: Thursday, November 17, 2005 1:23 PM
To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] VoIP Gateway ProvidersHi,Can anyone recommend a good reputable VoIP gateway service provider that I
can use with my Asterisk server in wa.us? All I can seem to find is VoIPservice directly to the desk. I'd prefer a provider that can provideDID-type services, because that is my big selling point to the company.
Thanks,Jeff RamseyMIS AdministratorTubafor Mill, Inc.[EMAIL PROTECTED]360.269.1650--No virus found in this incoming message.
Checked by AVG Free Edition.Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005--No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.362 / Virus Database: 
267.13.3/173 - Release Date: 11/16/2005___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://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 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

Re: [Asterisk-Users] PSGW 2.2 Skype gateway?

2005/11/18, Francesco Peeters [EMAIL PROTECTED]:
  I think we will need to wait until someone hacked the skype
  protocol... If there is someone interested on doing it.
 
 

 Which is going to be a pain, as it is encrypted...  :-(

 Reverse-engineering may be the best option, and that is:
 1) Not trivial
 2) Not always legal

 /me sighs...

 I'll try getting my friends on to VoipBuster instead!  ;-)

¡Right!. Thiis is the explanation of why there is not an easy way to
link skype with asterisk.

But there is people still thinking on doing it, because switchng from
skype to voipbuster is as sifficult as switching from
micro$oft-messenger to jabber: there is no reason for not doing it,
but people doesn't.

On other way, I must accept that skype codec has a very good compression.
--
Alejandro Vargas
___
--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

Re: [Asterisk-Users] Re: Can anyone explain reason for this echo

2005-11-18 Thread Andrew Kohlsmith

On Friday 18 November 2005 00:30, Eric Bishop wrote:
 I purchased the following item:
 http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html

 As you can see not a very highly spec'd product but does the job well.

Perhaps not highly specc'd but with tail lengths of 64ms bidirectional or 
128ms unidirectional, it's already more capable than the software cancellers 
(16ms unidirectional with echocancel=128) and I believe that the VPM isn't 
all that much better, but I'm not 100% sure now that I can't find the specs 
on it.

 I don't accept the fact that mine is a special case. In fact if anything it
 should be better than most other scenarios as we are using Tier 1 hardware
 (all HP), Digium Rev 2 firmware and our rack is about 10 metres from the
 CO.

None of that really matters -- it's the overall disance from your RJ48 to the 
far end's phone that determines the TDM delay, and delays in your 
motherboard's PCI implementation that cause echo.   By far mostly the latter.

-A.
___
--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

Re: [Asterisk-Users] Re: chan_bluetooth

2005/11/17, José Luis Gómez [EMAIL PROTECTED]:
 Hello Victor.
 I had the same problem, but when I compile a new version of
 chan_bluetooth (ones form august) it works.
 Try to download and compile a new version. I`m using asterisk 1.0.9.

Is not there a way to avoid the bluetooth part by using an usb
connection to the phone?

--
Alejandro Vargas
___
--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

Re: [Asterisk-Users] create my own soft Phone

2005-11-18 Thread Zoa



http://iaxclient.sourceforge.net/



ram wrote:



Hi

i would like to create my own soft Phone
for my local office use

can any one guide me the URL for the same
of source Soft Phone
or resources to create

ram



___
--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





signature.asc
Description: OpenPGP digital signature
___
--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

Re: [Asterisk-Users] PSGW 2.2 Skype gateway?

2005-11-18 Thread Patrick

On Fri, 2005-11-18 at 12:56 +0100, Alejandro Vargas wrote:
[snip]
 On other way, I must accept that skype codec has a very good compression.

Iirc they use iLBC Wideband which is 16KHz and does not work with
Asterisk which uses 8KHz. I'm not an expert though so I might have
misunderstood.

Regards,
Patrick
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] wcfxo loads correclty after issuing twice the command ztcfg -vvvv !!

2005-11-18 Thread Bukoka Budoka


Hi to all,

when i issue the ztcfg command for the first time i get the message 
Changing signalling on channel 1 from Unused to FXS Kewlstart.


When i issue it for the second time i get the normal message 1 channels 
configured.


Has anyone any ideas of why not to have the normal behavior on the first 
place (i mean without passing from Unused to FXS Kewlstart and then to 1 
channels configured. ???


---

[EMAIL PROTECTED] ~]# ztcfg -vvv

Zaptel Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

Changing signalling on channel 1 from Unused to FXS Kewlstart


[EMAIL PROTECTED] ~]# ztcfg -vvv

Zaptel Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

My Zaptel is as follows:

fxsks=1
loadzone=us
defaultzone=us
--

My zapata is as follows:

[channels]
context=outgoing
;switchtype=national
signalling=fxs_ks
;rxwink=300  ; Atlas seems to use long (250ms) winks
;flash=1
usecallerid=no
hidecallerid=yes
callwaiting=no
usecallingpres=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=3.0
txgain=-1.0
group=1
callgroup=1
pickupgroup=1
immediate=yes
cidsignalling=bell
channel = 1


The zaptel driver as well as the wcfxo driver is loaded in rc.local:

modprobe zaptel
modprobe wcfxo
/sbin/ztcfg -v
/usr/sbin/asterisk -vvv  /var/log/asterisk/status-log 

thank you all,

Budoka.

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.com/


___
--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

Re: [Asterisk-Users] Newbie question. (Long)


Hi All:

I've been through the compile/install procedure pointed out by Vassil: I 
still crash on startup. Can anyone else give me some pointers, please?


Roger

Roger Hill wrote:


Thanks Vassil - I'll try those pointers and report back.

Roger

Vassil Kolarov wrote:


Hi Roger,

Following this instructions:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

I was able to install and run Asterisk several times without problems.

See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora

Regards,
Vassil Kolarov
www.ittconsult.com


Roger Hill wrote:


Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
downloaded the binary package.


Now I'm trying to put the working installation on my production 
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried the 
tarball download of 1.2.0.rc2, ran make OK, then make install, make 
samples.
When I tried to run Asterisk, I got (immediately) Illegal 
Instruction.

Tried on my FC4 laptop, worked just fine.
Concluded I needed FC4, so upgraded the server yesterday. Six hours 
later...

Reran make clean, make...
Same problem.
Then tried 1.2.0; same problem.
Then tried 1.0.9; same problem.
Finally removed everything to do with asterisk, pulled dowm 1.2.0 
tar ball again, and re-installed.

Same old problem, illegal instruction.

I did an strace, which follows. I don't know enough to decide what 
the strace is telling me. (The missing /etc/ld.so.preload is also 
missing on the FC4 laptop which works, so I concluded that that was 
not the problem.)


Any help much appreciated.

Regards
Roger

[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
brk(0)  = 0x8773000
access(/etc/ld.so.preload, R_OK)  = -1 ENOENT (No such file or 
directory)

open(/etc/ld.so.cache, O_RDONLY)  = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
close(3)= 0
open(/lib/libdl.so.2, O_RDONLY)   = 3
read(3, 
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213..., 512) = 
512

fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000

close(3)= 0
open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
read(3, 
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306..., 512) = 
512

fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xb7f84000
old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000


close(3)= 0
open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
read(3, 
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343..., 512) = 
512

fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000

close(3)= 0
open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22..., 
512) = 512

fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000

close(3)= 0
open(/lib/libresolv.so.2, O_RDONLY)   = 3
read(3, 
\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323..., 512) = 
512

fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC, 
MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE, 
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000

close(3)= 0
open(/lib/libssl.so.5, O_RDONLY)  = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26..., 
512) = 512

fstat64(3,

[Asterisk-Users] In France asterisk never detect hang up. Why ?

2005-11-18 Thread asterisk user dupont

Hello.

I am sorry my english is not good at all.

When i have a call from a fxo port of a tdm400p, asterisk waits one
minute before detecting that the caller has hang up his phone.

I have in my extension conf :
answer
background  (the prompt is 40 second long)
dial (on fxs port)  confgured for 30 seconds ringing.

if the caller hang up at the begining of the background prompt,
asterisk waits until he make ring the phone on the dial command for
the all 30 secondes before detecting the hang up.

Do you know if there is a way to repair that ?

here is what i see on asterisk when the caller hang up IMMEDITALY
after the test prompt begins :

*CLI -- Starting simple switch on 'Zap/4-1'
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing NoOp(Zap/4-1, 0675458745) in new stack
-- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack
-- Response timeout set to 20
-- Executing BackGround(Zap/4-1, barge) in new stack
-- Playing 'test' (language 'fr')
-- Executing Dial(Zap/4-1, Zap/2|0675458745|30) in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/4-1
-- Attempting native bridge of Zap/4-1 and Zap/2-1
-- Hungup 'Zap/2-1'
  == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1'
-- Executing Hangup(Zap/4-1, ) in new stack
  == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'


In my zapata.conf i have :

language=fr
default=fr
relaxdtmf=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
cidsignalling=v23
usecallerid=yes
group = 1
context=reseau
signalling=fxs_ks
callprogress=yes
busydetect=yes
callerid=asreceived
busycount=5
pulse=yes

In my zaptel.conf i have :

loadzone=fr
defaultzone=fr
fxoks=1-3
fxsks=4


If anyone can see what is wrong he will really help me.

thank you.
___
--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

Re: [Asterisk-Users] /spool/outgoing delays

2005-11-18 Thread Leif Neland

 Original Message 
From: Chris Cahill [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 17, 2005 1:15 PM
Subject: [Asterisk-Users] /spool/outgoing delays

Hi,

I have a rather interesting problem with my Asterisk setup at the
moment, and was wondering if anybody could shed any light on it!

The system is initiated by placing a call file into
/var/spool/asterisk/outgoing. This file calls asterisk, so it is
calling itself.

The process then goes on to call a few agi scripts, and ends up
creating another file (via agi) in the outgoing directory, this one
being the one that calls the outside world.

Are you *creating* the file in the /outgoing directory?
You should create it somewhere else and move it into /outgoing, to prevent 
asterisk to find an incomplete file.

Leif

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problems with Read() in outgoing calls

2005-11-18 Thread Chris Cahill

I have used Read() in many inbound context (ie. when a user dials me).

I have an outbound call between asterisk and a user, initiated by a call 
file
in the outgoing directory, but Read() does not seem to take any input in
this situation.

Is there anyway of getting round this?

Scouse

-- 
=

...the UK's fastest growing mobile messaging provider...
(internet works July 2004)

(aq) Limited

194 Harrogate Road, Leeds, LS7 4NZ
http://www.aqcorporate.com
Sales: 08707 449227
Fax: 08707 449228
PGP public key http://aql.co.uk/A-Q.txt
Subject to our standard terms, found at
http://aql.co.uk/aq.php?p=terms

=

CONFIDENTIALITY

This e-mail is intended only for the use of the addressees named
above and may be confidential or legally privileged.
If you are not an addressee you must not read it and must not use
any information contained in nor copy it nor inform any person other
than (aq) limited or the addressees of its existence or contents.
If you have received this email and are not a named addressee, please
delete it and notify the (aq) limited customer service department via
[EMAIL PROTECTED] 



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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: /spool/outgoing delays

2005-11-18 Thread Chris Cahill


 The process then goes on to call a few agi scripts, and ends up
 creating another file (via agi) in the outgoing directory, this one
 being the one that calls the outside world.

 Are you *creating* the file in the /outgoing directory?
 You should create it somewhere else and move it into /outgoing, to prevent 
 asterisk to find an incomplete file.

 Leif

Leif,

Thanks for your suggestion, but yes I am creating it elsewhere and moving it 
in.

Regards,

Chris 



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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: ip phone

2005-11-18 Thread Doug Meredith

stevanus [EMAIL PROTECTED] wrote:

Maybe grandstream budgetone 100 series will fulfill your requirement.
It's very good for such a cheap sub-50 phone.

We have two of these and they are the VoIP equivalent of a $10 K-Mart
phone.  I won't even use them in my house, much less the office.

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

___
--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

Re: [Asterisk-Users] Newbie question. (Long)

2005-11-18 Thread Rich Adamson

Asterisk runs just fine on fc3. Best guess on your problem is that you've
got come default config parameters in /etc/asterisk directory that it is
not liking at all. You might try starting asterisk with 'asterisk -cvd'
and watch the output for errors.


 Hi All:
 
 I've been through the compile/install procedure pointed out by Vassil: I 
 still crash on startup. Can anyone else give me some pointers, please?
 
 Roger
 
 Roger Hill wrote:
 
  Thanks Vassil - I'll try those pointers and report back.
 
  Roger
 
  Vassil Kolarov wrote:
 
  Hi Roger,
 
  Following this instructions:
 
  http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
 
  I was able to install and run Asterisk several times without problems.
 
  See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
 
  Regards,
  Vassil Kolarov
  www.ittconsult.com
 
 
  Roger Hill wrote:
 
  Hi all :
 
  My first posting to the group - please be gentle!
 
  I've been messing with Asterisk for a couple of weeks now.
  1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
  downloaded the binary package.
 
  Now I'm trying to put the working installation on my production 
  server along with HTTP etc.
  ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
  2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
  GNU/Linux).
 
  That box, until yesterday, was running Fedora core 3. I tried the 
  tarball download of 1.2.0.rc2, ran make OK, then make install, make 
  samples.
  When I tried to run Asterisk, I got (immediately) Illegal 
  Instruction.
  Tried on my FC4 laptop, worked just fine.


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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Asterisk en france

2005-11-18 Thread asterisk user dupont

Bonjour,

J'ai changé en tel que ci dessous, et j'ai toujours le même probleme.
Il detect toujours pas le raccroché.

I have changed to this new file, and i still have the same problem.
Still not detecting hang up.

[channels]
language=fr
default=fr
relaxdtmf=yes
rxgain=0.0
txgain=0.0
usecallerid=yes
cadence=250,1500,1500,3000,1500,3000
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
group = 1
context=reseau
signalling=fxs_ks
callprogress=no
busydetect=yes
callerid=asreceived
busycount=3
pulse=yes
channel = 4
group = 2
callgroup=2
pickupgroup=2
context=local
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
useincomingcalleridonzaptransfer=yes


2005/11/18, Dave Cotton [EMAIL PROTECTED]:
 Chez moi j'ai

 [channels]
 language=en
 callwaiting=yes
 callwaitingcallerid=yes
 callprogress=no
 busydetect=yes ;changed 17.03.04 from no
 busycount=7   ; added as above
 immediate=no
 usecallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 echocancel=yes
 echocancelwhenbridged=yes
 echotraining=800
 rxgain=0.0
 txgain=0.0
 musiconhold=default
 ;faxdetect=incoming
 cadence=250,1500,1500,3000,1500,3000

 Chez toi je me demande pourquoi

 cidsignalling=v23
 callprogress=yes
 pulse=yes

 mes penses.


 --
 Dave Cotton [EMAIL PROTECTED]


___
--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

Re: [Asterisk-Users] Re: ip phone

2005-11-18 Thread Rich Adamson


 Maybe grandstream budgetone 100 series will fulfill your requirement.
 It's very good for such a cheap sub-50 phone.
 
 We have two of these and they are the VoIP equivalent of a $10 K-Mart
 phone.  I won't even use them in my house, much less the office.

Might be carefull with assumptions in this area... depending upon where
you are from and what type of service one is accustomed to using (or
receiving), the term quality has as many interpretations as there
are countries (or counties in some cases) in this world. Some would 
consider the 100 series as a significant improvement over what they 
currently have for service, while many others would consider it close 
to the bottom of the stack of sip phones.

I'm not trying to defend anyone's opinion or propose alternatives.


___
--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

Re: [Asterisk-Users] Newbie question. (Long)


Rich: Thanks.

I tried that, with and without any config files in /etc/asterisk. It 
still falls over instantly, no messages other than 'Illegal Instruction'.
Asterisk is running on other machines for me quite happily, but just 
does not want to play nice on this box.


I'm sure I'm doing something silly, but for the life of me cannot see 
what it is.


It does not get as far as writing anything to any log files in 
/var/log/asterisk.


Roger

Rich Adamson wrote:


Asterisk runs just fine on fc3. Best guess on your problem is that you've
got come default config parameters in /etc/asterisk directory that it is
not liking at all. You might try starting asterisk with 'asterisk -cvd'
and watch the output for errors.


 


Hi All:

I've been through the compile/install procedure pointed out by Vassil: I 
still crash on startup. Can anyone else give me some pointers, please?


Roger

Roger Hill wrote:

   


Thanks Vassil - I'll try those pointers and report back.

Roger

Vassil Kolarov wrote:

 


Hi Roger,

Following this instructions:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

I was able to install and run Asterisk several times without problems.

See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora

Regards,
Vassil Kolarov
www.ittconsult.com


Roger Hill wrote:

   


Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
downloaded the binary package.


Now I'm trying to put the working installation on my production 
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried the 
tarball download of 1.2.0.rc2, ran make OK, then make install, make 
samples.
When I tried to run Asterisk, I got (immediately) Illegal 
Instruction.

Tried on my FC4 laptop, worked just fine.
 




___
--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

 



--

Roger Hill  07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.


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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Sipura SPA-841 Second Line Help

2005-11-18 Thread Dave Morrow

Title: Sipura SPA-841 Second Line Help

Hi all, I recently purchased Sipura SPA-841 phones for a group of users. While the phones are functioning great, I am having some troubles configuring one aspect. Hopefully someone will know what I am doing wrong.

On each of the phones, I have configured Line 1 as a private line. That's working fine.

My requirement is to have an extension 9000 ring on all of the phones' second line.

I've configured this extension in asterisk (extensions.conf and sip.conf) as I would any other extension.

Inside the SPA-841 interface, I configured Ext 2 with the appropriate SIP information and set the line appearance to Shared

In the Phone tab, I set the line appearance to shared.

With all this configured. It doesn't work. L2 on each of the phones simply flashes red and does not work.

Any help would be greatly appreciated!

David A. Morrow

Technical Systems Lead

Autodata Solutions Company

[EMAIL PROTECTED]

http://www.autodata.net

* PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE *

NEW !!! Tel: (519) 963-3020

Fax: (519) 451-6615

Poor planning on your part does not necessarily constitute an emergency on my part!

This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

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

Re: [Asterisk-Users] realtime callerid

2005-11-18 Thread Are

I don't think this is working with static or Realtime. I just tested it
both ways on two servers. No luck. voip-info.org is saying the same.

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

restrictcid: (yes/no) To have the callerid restricted - sent as ANI; use this to hide the caller ID.  This does not seem to work.


 If you are using Realtime it is easy to change the info in the
database so this can't be a big problem. You can contact me off list if
you want more information. If you want to prevent the original Caller
id from beeing lost you can store that in another field in your
database.
Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultantshttp://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP
AstBill DEMO: http://demo.astbill.com
___
--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

Re: [Asterisk-Users] SER Asterisk combination to get around NAT

2005-11-18 Thread Simone Cittadini


Stuart Hirst ha scritto:


Has anyone successfully used SER and Asterisk together on the same
server to get around NAT traversal issues.

I have looked at many of the NAT traversal topics which either involve
commercial products and significant costs or solutions such as STUN or
proprietary systems such as xten.

 

I've installed ser + mediaproxy + asterisk without much trouble 
following the docs you find at www.onsip.org

___
--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

Re: [Asterisk-Users] Problem switching from external ISDN-2 to PBX ISDN-2

2005-11-18 Thread Lars Dybdahl

My extensions.conf is basically:

[incoming]
exten = s,1,Answer()
exten = s,2,Background(M800)
exten = s,3,WaitExten(5)
exten = s,4,Dial(Zap/2/12341234)

On the external ISDN-2 connection I get all incoming calls on Zap/1. I
use Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o and this is my zapata.conf:

[channels]
language=da
switchtype=euroisdn
;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier
to unknown.
pridialplan=local
prilocaldialplan=local


signalling = bri_cpe_ptmp
;signalling = fxs_ks
rxwink=300

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;rxgain=0.0
txgain=20
nationalprefix = 0
internationalprefix = 00
faxdetect=incoming
group=0
callgroup=1
pickupgroup=1
immediate=yes
context=isdnincoming
channel = 1-2


On 11/18/05, amer karim [EMAIL PROTECTED] wrote:
 Hi;

 What is ur extension.conf?
 do u use overlapdial??


 2005/11/18, Lars Dybdahl [EMAIL PROTECTED]:
 
  I have a system that works perfectly using zaphfc on an external
  ISDN-2 connection. When I move this to a PBX-based ISDN-2 connection,
  it still receives calls, but is unable to dial out.
 
  In Denmark, we have no long distance calls, but only national calls
  (8-digit numbers) or international calls (00 countrycode etc.).
  Normally, incoming national calls would be reported as 8-digit
  numbers, but when I connect my asterisk to the PBX, it is preceded by
  three zeroes (00012345678 instead of 12345678). This seems to indicate
  a problem, too, since you normally only dial one zero on a PBX to get
  an outside line.
 
  Any ideas?
 
  Lars.
  ___
  --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
 



 --
  cordialement
 Karim AMER
 ___
 --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 list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] 'ztmonitor' stopped working after using 'fxotune'

2005-11-18 Thread Chuck Bunn


Hi,

I cannot get 'ztmonitor' to run anymore after I ran 'fxotune'. I get the 
following error:


[EMAIL PROTECTED] ~]# cd /usr/src/zaptel
[EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -
Unable to open /dev/dsp: No such file or directory
Cannot open audio ...
[EMAIL PROTECTED] zaptel]#

I am using Asterisk 1.2 running on Fedora 4. I have also tried 
recompiling the Zaptel drivers but this did not fix the problem.


Thanks
___
--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

Re: [Asterisk-Users] suggestions for hard phones?

2005-11-18 Thread Health Masters

We currently use the Grandstream GXP-2000 . Although nice phones I would 
not get them if you were to use with out headsets.
The handsets are cheap and pick up noices around the phone. You actually 
get a better sounding call if you put your hand over the mouth piece and 
then talk with the phone near your chest.


I also heard something about the headset jack being non standard??

John Fraser wrote:


Hi all,

I am looking for SIP hard phones to use in a call center.
The feature that I need the most is quick change of logon credentials as we 
run 3 shifts. each agent will have their own extension number and password. 
any suggestions would be greatly appreciated.


thank you
John Fraser 
___

--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 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

Re: [Asterisk-Users] Newbie question. (Long)

2005-11-18 Thread Rich Adamson

Well... the next best guess is the binary package that you downloaded
has some dependencies that are not on your system, or, the package
simply wasn't intended for your distro (for one reason or another).

Does the system have a developement environment that would allow you
down download the cvs source and compile it?



 Rich: Thanks.
 
 I tried that, with and without any config files in /etc/asterisk. It 
 still falls over instantly, no messages other than 'Illegal Instruction'.
 Asterisk is running on other machines for me quite happily, but just 
 does not want to play nice on this box.
 
 I'm sure I'm doing something silly, but for the life of me cannot see 
 what it is.
 
 It does not get as far as writing anything to any log files in 
 /var/log/asterisk.
 
 Roger
 
 Rich Adamson wrote:
 
 Asterisk runs just fine on fc3. Best guess on your problem is that you've
 got come default config parameters in /etc/asterisk directory that it is
 not liking at all. You might try starting asterisk with 'asterisk -cvd'
 and watch the output for errors.
 
 
   
 
 Hi All:
 
 I've been through the compile/install procedure pointed out by Vassil: I 
 still crash on startup. Can anyone else give me some pointers, please?
 
 Roger
 
 Roger Hill wrote:
 
 
 
 Thanks Vassil - I'll try those pointers and report back.
 
 Roger
 
 Vassil Kolarov wrote:
 
   
 
 Hi Roger,
 
 Following this instructions:
 
 http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
 
 I was able to install and run Asterisk several times without problems.
 
 See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
 
 Regards,
 Vassil Kolarov
 www.ittconsult.com
 
 
 Roger Hill wrote:
 
 
 
 Hi all :
 
 My first posting to the group - please be gentle!
 
 I've been messing with Asterisk for a couple of weeks now.
 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
 downloaded the binary package.
 
 Now I'm trying to put the working installation on my production 
 server along with HTTP etc.
 ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
 GNU/Linux).
 
 That box, until yesterday, was running Fedora core 3. I tried the 
 tarball download of 1.2.0.rc2, ran make OK, then make install, make 
 samples.
 When I tried to run Asterisk, I got (immediately) Illegal 
 Instruction.
 Tried on my FC4 laptop, worked just fine.
   
 
 
 
 ___
 --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
 
   
 
 
 -- 
 
 Roger Hill07739 707 180
 Perseverance is the hard work you do after you get
 tired of doing the hard work you already did.
 
 
 ___
 --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
 

---End of Original Message-


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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Examples of LIMIT_CONNECT_FILE and other LIMIT_XX Options

2005-11-18 Thread Obelix


I want to play a sound on the connection of a call using the LIMIT_CONNECT_FILE
option but can't find any examples.

Does anyone have any examples? Examples of the usage of the other LIMIT_xx
options would also be appreciated.

Obelix



This message was sent using IMP, the Internet Messaging Program.

___
--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

Re: [Asterisk-Users] Sipura SPA-841 Second Line Help

2005-11-18 Thread Jerry Jones

You cannot have multiple devices registering with the same name in  
asterisk.

Only the most recent to register will actually receive the call.
Create a new registration ie 9001, 9002, etc on each of the phones  
then have 9000 ring all of them.


On Nov 18, 2005, at 7:32 AM, Dave Morrow wrote:

Hi all, I recently purchased Sipura SPA-841 phones for a group of  
users.  While the phones are functioning great, I am having some  
troubles configuring one aspect.  Hopefully someone will know what  
I am doing wrong.


On each of the phones, I have configured Line 1 as a private line.  
That's working fine.
My requirement is to have an extension 9000 ring on all of the  
phones' second line.
I've configured this extension in asterisk (extensions.conf and  
sip.conf) as I would any other extension.
Inside the SPA-841 interface, I configured Ext 2 with the  
appropriate SIP information and set the line appearance to Shared


In the Phone tab, I set the line appearance to shared.

With all this configured.  It doesn't work.  L2 on each of the  
phones simply flashes red and does not work.


Any help would be greatly appreciated!


David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net

* PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL  
CHANGE *


NEW !!! Tel: (519) 963-3020
Fax: (519) 451-6615

 Poor planning on your part does not necessarily constitute an  
emergency on my part! 


This message has originated from Autodata Solutions. The attached  
material is the Confidential and Proprietary Information of  
Autodata Solutions. This email and any files transmitted with it  
are confidential and intended solely for the use of the individual  
or entity to whom they are addressed. If you have received this  
email in error please delete this message and notify the Autodata  
system administrator at [EMAIL PROTECTED]  
mailto:[EMAIL PROTECTED]


___
--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 list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] DNS SRV

2005-11-18 Thread Usman




Hi,

I need to run sip on non-standard port e,g 8881 and do not 
want user to define this port in clients like ata or softphone.
what I want, when a client sends a register request at sip 
server, the sip server should send him the port number OR is there a way 
using DNS SRV  

can any 1 help me out ?



___
--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

Re: [Asterisk-Users] Help with shell script for externnotify

2005-11-18 Thread Tzafrir Cohen

On Thu, Nov 17, 2005 at 09:32:37PM -0500, Tom Rymes wrote:
 Hi folks,
 
 I am working on a shell script that I can use with the externnotify  
 command in voicemail.conf. All is well and seems ready to rock,  
 except I can't figure out how to tell the script what e-mail address  
 to send the mail messages to. I warn you ahead of time that I am no  
 scripting guru.
 
 Basically, I have 14 after-hours mailboxes that all have different e- 
 mail addresses. I want this script to parse the mailbox number from  
 the original command ($2), and then somehow look that up mailbox's  
 address and send an e-mail. 

This is a mail routing issue. Why not configure the mail server
properly?

in voicemail.conf configure each extension's email to be
'NUM@localhost . Handle the forwarding using the mail server's aliases
file. This is typically /etc/aliases , but there are variations
depending on the specific MTA (postfix? sendmail? exim? qmail?) and on
the installation.

Run two daily cron jobs that change the contents of the aliases file.
Spefically with postfix you can give it a number of aliases files, and
thus the cron job can edit one file, run 'newaliases' and be done with
it.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--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

Re: [Asterisk-Users] Newbie question. (Long)


Rich:

Sorry if I did not make myself clear.

I was trying to give some history, which is where the downloaded package 
came from.


On this box (FC4), I am currently downloading the 1.2.0 source from 
asterisk.org (but not the CVS), and trying to compile and build from 
scratch.


The build seems fine - if it will help I can post the output from the 
makes - but the built executable just crashes. I have done the same 
thing on another FC4 box (my laptop) without any problems.


Doees that help at all? (And many thanks for the help, BTW)
Roger

Rich Adamson wrote:


Well... the next best guess is the binary package that you downloaded
has some dependencies that are not on your system, or, the package
simply wasn't intended for your distro (for one reason or another).

Does the system have a developement environment that would allow you
down download the cvs source and compile it?



 


Rich: Thanks.

I tried that, with and without any config files in /etc/asterisk. It 
still falls over instantly, no messages other than 'Illegal Instruction'.
Asterisk is running on other machines for me quite happily, but just 
does not want to play nice on this box.


I'm sure I'm doing something silly, but for the life of me cannot see 
what it is.


It does not get as far as writing anything to any log files in 
/var/log/asterisk.


Roger

Rich Adamson wrote:

   


Asterisk runs just fine on fc3. Best guess on your problem is that you've
got come default config parameters in /etc/asterisk directory that it is
not liking at all. You might try starting asterisk with 'asterisk -cvd'
and watch the output for errors.




 


Hi All:

I've been through the compile/install procedure pointed out by Vassil: I 
still crash on startup. Can anyone else give me some pointers, please?


Roger

Roger Hill wrote:

  

   


Thanks Vassil - I'll try those pointers and report back.

Roger

Vassil Kolarov wrote:



 


Hi Roger,

Following this instructions:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

I was able to install and run Asterisk several times without problems.

See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora

Regards,
Vassil Kolarov
www.ittconsult.com


Roger Hill wrote:

  

   


Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
downloaded the binary package.


Now I'm trying to put the working installation on my production 
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried the 
tarball download of 1.2.0.rc2, ran make OK, then make install, make 
samples.
When I tried to run Asterisk, I got (immediately) Illegal 
Instruction.

Tried on my FC4 laptop, worked just fine.


 


___
--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



 


--

Roger Hill  07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.


___
--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

   



---End of Original Message-


___
--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

 



--

Roger Hill  07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.


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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Remove older version of Asterisk

2005-11-18 Thread gc





Ihave an older version (0.9.0)of 
Asterisk on my linux box. Do I need to remove it before I install version 1.2? 
How do I remove it? Does Asterisk make file contain the uninstall process? Or I 
have to manully remove all the directory structure. 

Gary
___
--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

Re: [Asterisk-Users] gpx-2000 early dial support

2005-11-18 Thread Leif Neland

 Original Message 
From: Louis-David Mitterrand [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 18, 2005 10:10 AM
Subject: [Asterisk-Users] gpx-2000 early dial support

The gxp-2000's lack of a dialplan (or did I miss it?) led me to
activate its early dial option to avoid pressing Send after
dialing. Thus the dialplan is controlled by asterisk.

It creates an extension matching problem:

exten = _00[1-9].,1,Macro(dialcapi)

If I dial 0012 the extension is matched immediately. Is there a way to
ask asterisk to wait a few seconds for more digits?

You seem to contradict yourself.

You want to call a few seconds after the last digit.

Why implement it in asterisk, when the phone is capable of doing that by 
itself.

Let the phone decide when these few seconds has expired.
Remove the early dial again, and set the timeout in the phone.

My Grandstreams have 4 seconds digit timeout.

Leif

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk 1.2 - Windows Messenger ?

2005-11-18 Thread Robert Rozman


Hi,

I've found quite some docs on this, but many of them deprecated...

I'm curious what is the latest window messenger version that works as 
registered client to Asterisk... I've tried 4.7, but it registers only if I 
leave password empty.


Am I missing something or is there any better way to register and use 
Windows messenger with Asterisk ?


Any other sucessful experience with Windows Messenger and Asterisk ?

Thanks in advance,

regards,

Rob.

___
--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

Illegal Instruction on new FC4 install [was: Re: [Asterisk-Users] Newbie question. (Long)]

2005-11-18 Thread Tzafrir Cohen

On Fri, Nov 18, 2005 at 09:12:46AM +, Roger Hill wrote:
 Hi all :
 
 My first posting to the group - please be gentle!

Please use a more descriptive subject line. 

 
 I've been messing with Asterisk for a couple of weeks now.
 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), 
 downloaded the binary package.
 
 Now I'm trying to put the working installation on my production server 
 along with HTTP etc.
 ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 
 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 
 GNU/Linux).

 
 That box, until yesterday, was running Fedora core 3. I tried the 
 tarball download of 1.2.0.rc2, ran make OK, then make install, make samples.
 When I tried to run Asterisk, I got (immediately) Illegal Instruction.
 Tried on my FC4 laptop, worked just fine.
 Concluded I needed FC4, so upgraded the server yesterday. Six hours later...
 Reran make clean, make...
 Same problem.
 Then tried 1.2.0; same problem.
 Then tried 1.0.9; same problem.
 Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar 
 ball again, and re-installed.
 Same old problem, illegal instruction.

What CPU is it? cat /proc/cpuinfo

 
 I did an strace, which follows. I don't know enough to decide what the 
 strace is telling me. (The missing /etc/ld.so.preload is also missing on 
 the FC4 laptop which works, so I concluded that that was not the problem.)

Indeed it is not a problem.

 
 Any help much appreciated.
 
 Regards
 Roger
 
 [EMAIL PROTECTED] sbin]$ sudo strace ./asterisk

When you try to strace a process that may fork, try 'strace -f' instead.

Also: what does a simple 'asterisk -cddvv' give?

 execve(./asterisk, [./asterisk], [/* 27 vars */]) = 0
 uname({sys=Linux, node=coach.hillconsult.com, ...}) = 0
 brk(0)  = 0x8773000
 access(/etc/ld.so.preload, R_OK)  = -1 ENOENT (No such file or 
 directory)
 open(/etc/ld.so.cache, O_RDONLY)  = 3
 fstat64(3, {st_mode=S_IFREG|0644, st_size=48722, ...}) = 0
 old_mmap(NULL, 48722, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f85000
 close(3)= 0
 open(/lib/libdl.so.2, O_RDONLY)   = 3
 read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\250\213..., 
 512) = 512
 fstat64(3, {st_mode=S_IFREG|0755, st_size=16760, ...}) = 0
 old_mmap(0x8f8000, 12388, PROT_READ|PROT_EXEC, 
 MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8f8000
 old_mmap(0x8fa000, 8192, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x1000) = 0x8fa000
 close(3)= 0
 open(/lib/tls/i686/libpthread.so.0, O_RDONLY) = 3
 read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\334\306..., 
 512) = 512
 fstat64(3, {st_mode=S_IFREG|0755, st_size=103404, ...}) = 0
 old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, 
 -1, 0) = 0xb7f84000
 old_mmap(0x938000, 65980, PROT_READ|PROT_EXEC, 
 MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x938000
 old_mmap(0x945000, 8192, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xc000) = 0x945000
 old_mmap(0x947000, 4540, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x947000
 
 close(3)= 0
 open(/usr/lib/libncurses.so.5, O_RDONLY) = 3
 read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\200\343..., 
 512) = 512
 fstat64(3, {st_mode=S_IFREG|0755, st_size=985952, ...}) = 0
 old_mmap(0x5a0, 290348, PROT_READ|PROT_EXEC, 
 MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x5a0
 old_mmap(0x5a3e000, 36864, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3d000) = 0x5a3e000
 close(3)= 0
 open(/lib/tls/i686/libm.so.6, O_RDONLY) = 3
 read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320\22..., 
 512) = 512
 fstat64(3, {st_mode=S_IFREG|0755, st_size=213872, ...}) = 0
 old_mmap(0x8fe000, 139424, PROT_READ|PROT_EXEC, 
 MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x8fe000
 old_mmap(0x91f000, 8192, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x91f000
 close(3)= 0
 open(/lib/libresolv.so.2, O_RDONLY)   = 3
 read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\350\323..., 
 512) = 512
 fstat64(3, {st_mode=S_IFREG|0755, st_size=72956, ...}) = 0
 old_mmap(0x94b000, 71848, PROT_READ|PROT_EXEC, 
 MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x94b000
 old_mmap(0x959000, 8192, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0xd000) = 0x959000
 old_mmap(0x95b000, 6312, PROT_READ|PROT_WRITE, 
 MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x95b000
 close(3)= 0
 open(/lib/libssl.so.5, O_RDONLY)  = 3
 read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\26..., 
 512) = 512
 fstat64(3, {st_mode=S_IFREG|0755, st_size=230056, ...}) = 0
 old_mmap(0xaa8000, 228948, PROT_READ|PROT_EXEC, 
 MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xaa8000
 old_mmap(0xadd000, 12288, PROT_READ|PROT_WRITE,

Re: [Asterisk-Users] wcfxo loads correclty after issuing twice the command ztcfg -vvvv !!

2005-11-18 Thread Tzafrir Cohen

On Fri, Nov 18, 2005 at 01:18:50PM +0100, Bukoka Budoka wrote:
 Hi to all,
 
 when i issue the ztcfg command for the first time i get the message 
 Changing signalling on channel 1 from Unused to FXS Kewlstart.
 
 When i issue it for the second time i get the normal message 1 channels 
 configured.
 
 Has anyone any ideas of why not to have the normal behavior on the first 
 place (i mean without passing from Unused to FXS Kewlstart and then to 1 
 channels configured. ???

The default behaviour of ztcfg is to print nothing, unless there is an
error. You obviously want to know a bit more about what exactly it does,
which is why you ask it to tell you more details:

 
 ---
 
 [EMAIL PROTECTED] ~]# ztcfg -vvv

That is: ztcfg: pelase be very verbose

And in return it tells you not only that it sets the signalling to FXS,
but also that it wasn't set previously. A minor, technical detail.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: gpx-2000 early dial support

2005-11-18 Thread Louis-David Mitterrand

On Fri, Nov 18, 2005 at 03:30:32PM +0100, Leif Neland wrote:
 The gxp-2000's lack of a dialplan (or did I miss it?) led me to
 activate its early dial option to avoid pressing Send after
 dialing. Thus the dialplan is controlled by asterisk.
 
 It creates an extension matching problem:
 
 exten = _00[1-9].,1,Macro(dialcapi)
 
 If I dial 0012 the extension is matched immediately. Is there a way to
 ask asterisk to wait a few seconds for more digits?
 
 You seem to contradict yourself.
 
 You want to call a few seconds after the last digit.
 
 Why implement it in asterisk, when the phone is capable of doing that by 
 itself.
 Let the phone decide when these few seconds has expired.
 Remove the early dial again, and set the timeout in the phone.
 
 My Grandstreams have 4 seconds digit timeout.

Oh, I have missed that one.

So basically there is no dialplan in-phone, but a key timeout after
which the number will be sent to * if no Send or # key are pressed?



-- 
The Information Revolution will be fought on the command line.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] phone intergration

2005-11-18 Thread Stas Khromoy


hey folks

is there a way to integrate toshiba  dkt2010-sd into asterik network ?
___
--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

Re: [Asterisk-Users] Newbie question. (Long)

2005-11-18 Thread Vassil Kolarov


Roger,
Can you try with a fresh Fedora installation on this box?

Vassil

Roger Hill wrote:

Rich:

Sorry if I did not make myself clear.

I was trying to give some history, which is where the downloaded 
package came from.


On this box (FC4), I am currently downloading the 1.2.0 source from 
asterisk.org (but not the CVS), and trying to compile and build from 
scratch.


The build seems fine - if it will help I can post the output from the 
makes - but the built executable just crashes. I have done the same 
thing on another FC4 box (my laptop) without any problems.


Doees that help at all? (And many thanks for the help, BTW)
Roger

Rich Adamson wrote:


Well... the next best guess is the binary package that you downloaded
has some dependencies that are not on your system, or, the package
simply wasn't intended for your distro (for one reason or another).

Does the system have a developement environment that would allow you
down download the cvs source and compile it?



 


Rich: Thanks.

I tried that, with and without any config files in /etc/asterisk. It 
still falls over instantly, no messages other than 'Illegal 
Instruction'.
Asterisk is running on other machines for me quite happily, but just 
does not want to play nice on this box.


I'm sure I'm doing something silly, but for the life of me cannot 
see what it is.


It does not get as far as writing anything to any log files in 
/var/log/asterisk.


Roger

Rich Adamson wrote:

  
Asterisk runs just fine on fc3. Best guess on your problem is that 
you've
got come default config parameters in /etc/asterisk directory that 
it is
not liking at all. You might try starting asterisk with 'asterisk 
-cvd'

and watch the output for errors.






Hi All:

I've been through the compile/install procedure pointed out by 
Vassil: I still crash on startup. Can anyone else give me some 
pointers, please?


Roger

Roger Hill wrote:

 
  

Thanks Vassil - I'll try those pointers and report back.

Roger

Vassil Kolarov wrote:

   


Hi Roger,

Following this instructions:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3

I was able to install and run Asterisk several times without 
problems.


See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora

Regards,
Vassil Kolarov
www.ittconsult.com


Roger Hill wrote:

 
  

Hi all :

My first posting to the group - please be gentle!

I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, 
Kubuntu), downloaded the binary package.


Now I'm trying to put the working installation on my production 
server along with HTTP etc.
( 700MHz, 256MB ram, uname -a gives Linux 
coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 
EST 2005 i686 i686 i386 GNU/Linux).


That box, until yesterday, was running Fedora core 3. I tried 
the tarball download of 1.2.0.rc2, ran make OK, then make 
install, make samples.
When I tried to run Asterisk, I got (immediately) Illegal 
Instruction.

Tried on my FC4 laptop, worked just fine.
   


___
--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





--

Roger Hill07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.


___
--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

  


---End of Original Message-


___
--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 list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk 1.2 and music-on-hold question

2005-11-18 Thread Gary MacKay

Are there any new docs on getting moh to work with v1.2? I successfully, 
I think, upgraded my 1.09 box to the v1.2 and all seems to work great 
except for moh. The UPGRADE.txt file mentions there are some 
differences, but does not go into any detail. I had it working great on 
v1.09 using madplay.

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

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] OT: Softphone with Bluetooth support for *

2005-11-18 Thread Remco Barende

I have seen some options for road warriors to connect a DECT phone to a 
SIP device or use a WIFI VOIP phone when travelling but I was wondering if 
it would be possible to use a soft phone and a standard bluetooth headset 
to connect to *


Most newer laptops have bluetooth support built in and a bluetooth headset 
is lots cheaper than a WIFI VOIP phone, not to mention easier to carry!


Has anyone ever tried such a setup?
___
--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

Re: [Asterisk-Users] phone intergration

2005-11-18 Thread John Novack




Stas Khromoy wrote:


hey folks

is there a way to integrate toshiba  dkt2010-sd into asterik network ?

These Toshiba phones  work ONLY with the Toshiba DK series of Key 
Service Units.

You would be able to connect the KSU to Asterisk, via a number of methods
It also depends on which KSU you have if it can be done via a T1 port, 
or if you have to go in and out via analog.


IF you don't have the Toshiba KSU, then the answer is NO!

John Novack

___
--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

Re: [Asterisk-Users] create my own soft Phone